AudioCodes Gateways: Essentials & Configuration

AudioCodes Academy
https://www.audiocodes.com/services-support/audiocodes-academy
Course Objectives
• Installation and Configuration of AudioCodes equipment using various management methods
• Operation and Maintenance of AudioCodes equipment (Performing backups, version
updating, configuration changes)
• Conducting basic interoperability and support services
• Identify and isolate relevant configuration parameters for a variety of services including SIP
proxy, fax, modems, and DTMF transport and dialing
• Provision digital trunks
• Describe and demonstrate AudioCodes gateway functionality with regard to call processing
and routing for Tel to IP and IP to Tel call scenarios
• Troubleshooting and Debugging of AudioCodes equipment

2
Lessons & Course Time Table
Day 1 Day 2
Welcome & Course Description Overview Lab 2 – SIP Call Tests
AudioCodes Introduction Lab 3 – Call with Proxy
Introduction to Legacy Telephony FXO
Introduction to IP Telephony Lab 4 – FXO
AudioCodes Documentation Common Features
MediaPack Overview Lab 5 – Hunt Group
Assigning Networking Parameters Lab 6 – Number Manipulation
Management Maintenance MPs
Management Maintenance Mediants
Basic SIP Configuration
Basic Debugging Tools
Lab 1 – Basic Call

3
Lessons & Course Time Table
Day 3 Day 4
Mediant Family Lab 10 – Primary Rate Interface
Mediant Family Configuration Advanced Debugging Tools
Lab 7 – Digital Gateways Basic Lab 11 – Debugging Tools
Lab 8 – Alternative Routing Certification Exam
Lab 9 – Profiles Course Summary

4
Lesson 1

AudioCodes Introduction
AudioCodes in a glance

• Market leader in VoIP networking products

• Deployed in over than 100 countries in service provider and enterprise networks
• Recognized brand for quality & performance
• Global partnerships with leading telecom players
• Large Fortune 100 install base
• Over 600 employees, ~40% R&D
• More than 20 years of VoIP expertise
• Public since 1999 (NASDAQ:AUDC)

6
Global Presence and Support

• Worldwide presence:
• Headquarters: Israel
• North America: USA and Canada
• APAC: Japan, Singapore, Korea, China, India, Australia, Hong Kong
• EMEA: UK, France, Netherland, Germany, Russia, Italy, South Africa, Poland, Sweden
• CALA: Miami, Brazil, Mexico, Argentina, Colombia
• Global Distribution Network covering more than 100 countries
• Support Centers covering all time zones
• 3 Logistics Centers in North America, EMEA and APAC

7
Our Customers

• 49 of Fortune 100 enterprises are using AudioCodes technology

• Hundreds of multinational enterprises
• Energy, Finance and Insurance, Industrial engineering, Food, Commerce,
Government and Defense, Pharmaceuticals, High Tech, Automotive
• Thousands of mid-market customers
via Service Providers and resellers

8
Broadest Portfolio of Products

Management/Apps
Routing Manager OVOC CloudBond 365/CCE Apps

IP Phones
405 420 430 440 450

Pure SBC
Mediant 2600 Mediant 4000 Mediant 9000

Software SBC
Mediant VE Virtual Edition Mediant SE Software Edition

Hybrid SBC/Gateway
Mediant 500/L Mediant 800 Mediant 1000 Mediant 3000

Gateways/Adaptors
MP-2xx MP-1xx MP-124 MP1288
9
AudioCodes All-in-One Voice Solution

Covering all aspects of VoIP solutions

10
Professional Services

We are passionate about satisfaction and drive to empower our business partners to
sell, support our customers, and deliver global value-added logistics services

Installation & Remote

Network Readiness Planning & Design
Implementation Monitoring
Assessments

12

9 3

24x7 Technical Spares

Support Software Upgrades Management Channel training

11
AudioCodes Complete Network Life-cycle Model

• Full product life cycle

• Plan
• Determine the right solution and best practices for any project’s needs

• Implement
• Achieve smooth voice implementations with global physical installation
and configuration

• Operate
• Prompt technical support, efficient hardware replacement and ongoing
software and hardware upgrades
12
Operational Services – ACTS & CHAMPS

• ACTS: Direct Support – Tier 2 – 4 (9 x 5 or 24 x 7)

• CHAMPS: Back-to-Back Support – Tier 3 – 4 (9 x 5 or 24 x 7)
• Not including installation, configuration, and provisioning (which can be
purchased separately)
• Support available after AudioCodes products are implemented and in service
• Support is provided based on serial number entitlement check Extended
Hardware Warranty (RMA) included
• Software Maintenance and all S/W upgrades, patches, maintenance releases
and major version releases
• Certificate of Eligibility issued with each purchase

13
Technical Training – Career Certifications

• Two types of Certification Levels:

• ACA – AudioCodes Certified Associate

• Basic level certification
• Required for the installation and maintenance of AudioCodes devices

• ACP – AudioCodes Certified Professional

• Advanced level certification
• Required for the installation, maintenance and advanced troubleshooting
of all AudioCodes networking products in advanced customer scenarios
• Prerequisite: ACA certification and 6 months of field experience as ACA

14
Technical Training Website

15
AudioCodes Website - www.audiocodes.com

16
Lesson 2

Introduction to Legacy Telephony

Objectives

After completing this lesson you will be able to:

• Explain the basics on telephone networks

• Describe how digital signaling differs from analog signaling

• Explain the basic concept of voice over IP communications

• Describe the purpose of the Gateway in a VoIP network

18
The Telephone Network

• PSTN (Public Switched Telephone Network) is the generic name for the worldwide
public telephone network
• Comprises mainly telephone switches (exchanges) and telephones, networked to
form the worldwide telephone communications system
• Also known as POTS (Plain Old Telephone Network)

19
The Telephone Network

Hierarchical Network

Intercity
Trunk
Trunk Trunk

Subscriber Loop Subscriber Loop

Subscribers Subscribers

20
The Telephone Network

Central Office Central Office

Switch Switch

Trunk

Subscriber Subscriber
Loop Loop

Subscriber Subscriber
Telephone Set Telephone Set
21
Typical Analog Circuit

• The twisted pair wires from the central switch office to a subscriber's home is called
a subscriber loop
• The subscriber loop handles two types of information: signals and voice on the
same twisted pair

Central Office
Subscriber

Telephone Set
Switch

22
Loop Start Signaling

Transmitter Switch Ringer Ring Battery and

Generator Current
hook Detector

Loop

Receiver

23
On-Hook

• In the on-hook stage, the switch is open and there is no current flow

Telephone
Switch

Local Local
Loop Loop

24
Off-Hook

• When the handset is picked up (going off-hook), a switch on the phone closes the
connection between the two wires and a -48 VDC current is drawn from the central
office switch
• The switch determines that the current is being drawn and provides a dial tone so
the caller knows it is time to dial a number

Telephone
Switch

Dial Tone

Local Local
Loop Loop

25
Dialing

• Upon hearing the dial tone, the user pushes the number buttons, which are
connected to a tone generator inside the dial, which generates DTMF tones
• The Telephone Switch collects the DTMF digits and maps them to a physical
subscriber

Telephone
Switch
Dialing (DTMF
Transmission)

Local Local
Loop Loop

26
DTMF - Dual Tone Multi-Frequency

• DTMF is the common method of sending dialing information (replaced pulse

dialing)
• Each number is represented by two tones which are transmitted simultaneously on
the voice path
• Each row represents a low frequency and each column represents a high frequency

1209 1336 1477 1633

697 1 2 3 A

770 4 5 6 B

852 7 8 9 C

941 * 0 # D

27
Ringing

• The Telephone Switch applies an AC ringing voltage which causes the sound
mechanism of the Called Telephone to ring

Telephone
Switch

Ringback Tone Ringing Voltage

Local Local
Loop Loop

28
Conversation

• The transmitter (handset’s microphone) generates an electric current which varies

in response to the acoustic pressure waves produced by the voice
• The resulting variations in electric current are transmitted along the telephone line
to the other phone

Telephone
Switch

Voice Voice

Local Local
Loop Loop

29
Call Tear Down

• When a party "hangs up" (puts the handset on the cradle), the DC current ceases to
flow in that line, signaling to the telephone switch to disconnect the call
• The switch plays a fast busy tone to the remote party

Telephone
Switch

Fast Busy Tone

Local Local
Loop Loop

30
Call Progress Tones

• In Telephony, call progress tones are audible tones sent from the PSTN or a PBX to
the calling / called parties to indicate the status of phone calls.
• Examples of some tones:

Call Progress Tone Description

Indicates that the telephone exchange is working, has recognized an
Dial Tone off-hook, and is ready to accept digits

Assures the calling party that a ringing signal is being sent on the
Ringback Tone called party's line

Busy Tone Indicates to the calling party that the remote phone is occupied

Indicates that a person has dialed an invalid code, or that all trunks are
Reorder Tone (Fast Busy) busy and/or their calls are un-routable

Indicates that a calling subscriber has been placed "on hold" by a

Hold tone subscriber with PBX or other facilities

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Telephony Network (1)

415-577-3800

415-577-3801
Telephone Switch
(Central Office)
415-577-3700

415-577-3701

415-577-3722

415-577-3733

Company X
415-577-3760

415-577-3785

32
Telephony Network (2)

415-577-3800

415-577-3801
Telephone Switch
(Central Office)
415-577-3700

415-577-3701

415-577-3722

Digital Trunk
415-577-3733

Company X 415-577-37xx
415-577-3760 PBX

415-577-3785

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Digital Communication

• A digital trunk is a single communication path between two switches that is used to
carry many simultaneous voice conversations

Local Central Office Remote Central Office

34
Pulse Code Modulation (PCM)
• A method of encoding an audio signal in digital
format world
• A standard audio signal is encoded by taking
8000 analog samples per second
• Each sample is quantized with an 8 bit value,
resulting in a digital signal known as DS0, to be
transferred at a speed of 64 kbps
• A companding algorithm with 2 different
flavors is used to reduce the quantization error.
• Mu-Law mainly used in North America and
Japan
• A-Law used in Europe and the rest of the
world

35
Time Division Multiplexing (TDM)

• Uses time-division multiplexing

1 1
64 Kbps

2 2
64 Kbps

3 2 1
3 3
64 Kbps

Multiplexer Demultiplexer

. . . 32 . . . 32
64 Kbps

36
E1
• Data rate of 2.048 Mbps (full duplex)
• Split into 32 time slots
• Each time slot sends and receives an 8-bit sample 8000 times per second
(8 x 8000 x 32 = 2,048 Mbps)
• Ideal for voice telephone calls where the voice is sampled into an 8 bit number (PCM)
• One timeslot (TS0) is reserved for framing purposes
• One timeslot (TS16) is often reserved for signaling purposes
Maintenance 1 2 15 17 30

Signaling
Framing

Voice

Voice

Voice

Voice

Voice
and

... ...

0 1 2 15 16 17 31

2.048 Mbps
37
T1
• Data rate of 1.544 Mbps
• Split into 24 time slots each encoded in 64 Kbps streams
• 8 Kbps of framing information for synchronization
• 64,000 x 24 + 8000 = 1544 Mbps
• Timeslot (TS24) is often reserved for signaling purposes

Signaling 23 22 15 13

Framing
Voice

Voice

Voice

Voice
Voice

Voice
... ...

24 23 22 15 14 13 1

1,536 Mbps

38
Signaling Methods
• In-band signaling is the exchange of signaling (call control) information on the same B-channel that the
telephone call itself is using
• Examples: CAS (Channel Associated Signaling), Loop Start

Voice + Signaling Link

• Out-of-band signaling is the exchange of signaling that is done on a channel that is dedicated for the
purpose and separate from the channels used for the telephone call
• Examples: Common Channel Signaling (CCS) such as the D Channel (ISDN) and SS7

Signaling Link

Voice Link

39
ISDN

• Integrated Services Digital Network is an ITU-T term for integrated transmission of

voice, video and data on the digital public telecommunications network
• Two interfaces are available:
• PRI (Primary Rate Interface) mostly used to link PBXs and to connect a PBX to the PSTN.
Composed of 23 or 30 B-channels and one D-channel, all at 64 Kbps
• BRI (Basic Rate Interface) an ISDN interface typically used by smaller sites and customers.
Consists of a single 16 Kbps D-channel plus 2 B-channels for voice and/or data

40
ISDN (Q.931) Call Flow

Calling Party Called Party

ISDN Digital Trunk

Off hook, Dial Tone, Dialing

Setup

Call Proceeding

Ringback Tone Alerting

Ringing

Connect
Off hook

Voice Channel

On hook Disconnect

Release

Release complete

41
BRI
• The ISDN Basic Rate Interface (BRI) service offers two B-channels and one
D-channel (2B+D)
• B-channel service operates at 64 kbps and is meant to carry user data
• D-channel service operates at 16 kbps and is meant to carry control and signaling
information
1 2

Framing and
Voice Voice
Signaling

D B1 B2

144 Kb/s

42
Lesson 3

Introduction to IP Telephony
Objectives

After completing this lesson you will be able to:

• Explain the basic concept of voice over IP communications

• Explain the basic SIP Call Flow
• Identify the SIP Network Entities
• Follow a SIP trace signaling in a call set-up and tear-down
• Describe the purpose of the Gateway in a VoIP network

44
What is VoIP

• VoIP is a set of technologies that enables the transmission of voice traffic over IP-
based networks instead of the Plain Old Telephone System (POTS)

45
Circuit vs. Packet Switching

• Circuit Switching
• Traditional voice calls, running over the PSTN, are made using circuit switching,
where a dedicated circuit or channel is set up between two points before the users
talk to one another
• Packet Switching
• A data transmission technique in which data is separated into small 'packets', each
with its own routing information and then sent through a shared, often public,
network
• At the other end, the packets are reassembled into the original data format.
In this method bandwidth is only used when something is actually being transmitted

46
VoIP Architecture

IP Telephone IP Telephone
IP Network

Gateway Gateway

PSTN PSTN

Legacy Telephones Legacy Telephones

47
VoIP Protocol Stack

• VoIP is composed of two key components:

• The bearer (the actual voice being sent over the network) using the RTP / RTCP protocols
• The signaling (which are additional messaging necessary to control, establish and tear-
down the voice calls).
The most common signaling protocols are: SIP, H.323, MGCP and MEGACO

48
VoIP Protocol Stack

SDP
MGCP
MEGACO
SIP

H.323 audio
video
H.225
H.245 Q.931 RAS RTP RTCP

TCP UDP

IPv4 / IPv6
Data Link Layer
Physical Layer

49
Introduction to RTP

V P X CC M PT Sequence Number

Time Stamp RTP Basic Header

12 octets
Synchronization Source ID - SSRC

Contributing Source ID – CSRC (only when CC is not 0)

Extension Header – Optional (when bit X = 1)

Voice Bits

50
Voice Codecs
• A codec (Coder/Decoder) converts analog signals to a digital bitstream, and back into an
analog signal for transmission across IP networks.
• Codecs generally provide a compression capability to save network bandwidth. Some codecs
also support silence suppression, where silence is not encoded or transmitted

Codec Bit Rate (kbps)

G.711 PCM (A-Law / Mu-Law) 64

G.726 ADPCM 16, 24, 32 and 40

G.729 CS-ACELP 8
G.723.1 ACELP 5.3
G.723.1 MPC-MLQ 6.3

51
VoIP Challenges

• Delay
• Each component in the path adds delay (sender, network, receiver)
• ITU-T G.114 recommends 150 msec as the maximum desired delay to achieve high voice
quality.
• Jitter
• Defines variation in delay
• The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival
and before producing audio
• Packet loss
• Occurs either in bursts or due to a congested network
• Periodic loss in excess of 5-10% of all VoIP packets can significantly degrade voice quality
52
Delay

Sender Receiver

Start Talk
Network

Packet X Packet X Arrive Start Hear

Transmitted

Processing Network Transit Processing time

Delay Delay Delay

End-to-End Delay

53
Jitter

• Jitter (delay variation) caused when voice packets suffer different transit delays, causing
variation in arrival times at the receiver
• The jitter buffer collects voice packets, stores them and sends them to the voice processor in
evenly spaced intervals

P1 P2 P3 Sender

time

P1 P2 P3 Receiver

time

D1 D2 = D1 D3 = D2

54
Introduction to SIP

• The Session Initiation Protocol (SIP) is a signaling protocol for initiating, managing
and terminating voice and video sessions across packet networks.
• It is based on a client-server architecture in which clients initiate calls and servers
answer calls.
• It is defined by IETF and documented in RFC 3261.

55
SIP Network Entities

• SIP defines two basic classes of network entities:

• Terminals - also known as Endpoints

• Servers - such as Proxy Server, Registrar and Redirect Server

56
User Agent (UA)

• A SIP User Agent is an entity which initiates and terminates sessions by exchanging requests and
responses
• The User Agents consists of two components:
• User Agent Client UAC (originates requests)
• User Agent Server UAS (reply for requests)

Request
UAC Response UAS

Request
UAS Response
UAC

57
Basic SIP Call Flow

SIP Phone (UAC) SIP Phone (UAS)

Off-hook & Dialing

INVITE (SDP)

100 Trying
Ringing
180 Ringing
Ringback
200 OK (SDP) Off-hook

ACK

RTP-RTCP

On-hook BYE

200 OK

58
SIP Requests: Basic Methods

Method Description

INVITE Establishes media sessions between user agents

ACK Acknowledges final responses to INVITE requests

BYE Terminates an established media session

Terminates a pending call attempt. Has no effect on an
CANCEL
established call.
OPTIONS Queries the capabilities of endpoints or servers

Endpoint notifies its current IP address and the URI for

REGISTER
which it would like to receive calls

59
SIP Requests: Extended Methods

Method Description
INFO Mid-call signaling (DTMF, hook-flash, etc.)
REFER Call transfer
User agent establishes a subscription for the purpose of receiving
SUBSCRIBE
notifications
User agent conveys information about the occurrence of a particular
NOTIFY
event (such as MWI)
PRACK Acknowledges receipt of reliably transported provisional responses (1xx)

UPDATE Modifies the state of a session

MESSAGE Sends instant messages
PUBLISH Publishes events on a server (e.g.,: Presence information)

60
SIP Responses

Informational Redirection
Indicates status of call prior to completion Server has returned possible locations. The
client should retry requests at another server.
100 Trying
180 Ringing 300 Multiple Choices
181 Call is being forwarded 301 Moved Permanently
182 Call Queued 302 Moved Temporarily
183 Session Progress 380 Alternative Service

Success
Request has succeeded
200 OK
202 Accepted

61
SIP Responses (cont.)
Client Errors Server Failure
The request has failed due to an error by the The request has failed due to an error by the server.
client. The client may retry the request if The request may be retried at another server.
reformulated according to response
500 Server Internal Error
400 Bad Request 501 Not Implemented
401 Unauthorized 502 Bad Gateway
403 Forbidden 503 Service Unavailable
404 Not Found
405 Method not Allowed Global Failure
407 Proxy Authentication Required
415 Unsupported Media The request has failed. The request should not be
486 Busy Here tried again at this or another server.
600 Busy Everywhere
603 Decline
604 Doesn’t Exist Anywhere
606 Not Acceptable
62
SIP Addressing

• SIP requests and responses are sent to particular addresses known as Uniform
Resource Identifier (SIP URI)
• Typically, routing is performed according to the Request-URI and not according to
the To header
• Convention: user@host
• User can be: user name or Tel number
• Host can be: domain name or IP address

INVITE sip:[email protected];user=phone SIP/2.0

INVITE sip:[email protected];user=phone SIP/2.0
INVITE sip:[email protected];user=phone SIP/2.0

63
INVITE

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 10.33.6.100;branch=z9hG4bKac1725404454
Max-Forwards: 70
From: “Mike” <sip:[email protected]>;tag=1c1725402038
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Supported: em,100rel,timer,replaces,path,resource-priority
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.5.00A.043.001
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 1725394110 1725393989 IN IP4 10.33.6.100
s=Phone-Call
c=IN IP4 10.33.6.100
t=0 0
m=audio 6000 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

64
200 OK

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.33.6.100;branch=z9hG4bKac1725404454
From: <sip:[email protected]>;tag=1c1725402038
To: <sip:[email protected];user=phone>;tag=1c1534094691
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Supported: em,timer,replaces,path,resource-priority
Server: Audiocodes-Sip-Gateway-MP-118 FXS_FXO/v.5.00A.043.001
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 1534112064 1534111943 IN IP4 10.33.6.101
s=Phone-Call
c=IN IP4 10.33.6.101
t=0 0
m=audio 6000 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

65
General Main Header Fields

Header Description

Call-ID The Call-ID uniquely identifies a particular dialog between two UAs

Cseq Contains a decimal number that increases for each request

Contains the name and the address of the originator of the request. Also
From
contains a tag, used to identify a particular call
Contains the name and the address of the called party.
To The “To” header field isn’t used for routing; the Request-URI is used for
this purpose.
Indicates the path in terms of proxies taken by the request. Used to
Via prevent looping on requests and to ensure that responses take the same
path.

66
Other important Header Fields

Header Description
Maximum number of proxies that can forward the request. Used to avoid
Max-Forwards
looping on requests in a similar way to IP’s TTL

Contact Indicates one or more addresses to use in order to contact the user

Supported Lists all the capabilities of the device

Used by clients to tell servers about the options that they expect the
Require
server to support in order to process the request

Content-Type Provides information about the type of the message body

67
Session Description Protocol (SDP)

• Provides negotiation between two SIP UAs to allow them to agree on a media type
and format
• Contains information on the media to be replaced such as RTP payload types, IP
address and ports
• Carried in SIP message body

v=0
o=AudiocodesGW 1725394110 1725393989 IN IP4 10.33.6.100
s=Phone-Call
c=IN IP4 10.33.6.100
t=0 0
m=audio 6000 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

68
PRACK

• PRACK Provisional Response ACKnowledge

• Used for informational response acknowledgements
• PRACK supported for calling and called sides
• PRACK must be sent if 180 message includes the ‘Require: 100rel’ header

PRACK

69
Early Media

• Used to set a voice connection prior to the establishment of the call (before
200 OK is received).
• Mostly used for playing announcements or Ringback tone.
• Method used: 183-Session Progress with SDP (instead of 180)

EarlyMedia

70
SIP Servers - Proxy

• Receives a SIP request from a user and acts on the user’s behalf in forwarding or
responding to the request
• Typically has access to a database or a location service to help it process the
request (determining the next hop)
• Performs functions such as:
• Authentication
• Authorization
• Network access control
• Routing Proxy Server

SIP Request SIP Request

SIP Response SIP Response

Media Session

71
Call Flow with Proxy (case 1)

SIP UA SIP Proxy SIP UA

INVITE (SDP)
100 Trying
INVITE (SDP)

100 Trying

180 Ringing
180 Ringing
200 OK (SDP)
200 OK (SDP)

ACK

RTP-RTCP

BYE

200 OK

72
INVITE after Traversing the Proxy

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP myproxy.mynet.com:5060;branch=z9hG4bK2d4790.1 added
Via: SIP/2.0/UDP 10.33.6.100;branch=z9hG4bKac1725404454
Max-Forwards: 69 modified
From: “Mike” <sip:[email protected]>;tag=1c1725402038
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Supported: em,100rel,timer,replaces,path,resource-priority
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.5.00A.043.001
Content-Type: application/sdp
Content-Length: 260

SDP description follows

73
Call Flow with Proxy (Case 2)

SIP UA SIP Proxy SIP UA

INVITE (SDP)
100 Trying
INVITE (SDP)

100 Trying

180 Ringing
180 Ringing
200 OK (SDP)
200 OK (SDP)
ACK
ACK

RTP-RTCP

BYE BYE
200 OK
200 OK

74
INVITE after Traversing the Proxy

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP myproxy.mynet.com:5060;branch=z9hG4bK2d4790.1 added
Via: SIP/2.0/UDP 10.33.6.100;branch=z9hG4bKac1725404454
Max-Forwards: 69 modified
Record-Route: <sip:myproxy.mynet.com;lr> added
From: “Mike” <sip:[email protected]>;tag=1c1725402038
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Supported: em,100rel,timer,replaces,path,resource-priority
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.5.00A.043.001
Content-Type: application/sdp
Content-Length: 260

SDP description follows

75
SIP Servers - Registrar

• A server that accepts SIP REGISTER requests from users.

REGISTER requests provide the server with an address at which the user can be
reached
• The registration server creates a temporary binding between the Address of Record
(AOR) URI in the “To” and the device URI in the Contact header
• The AOR identifies the user, while the Contact identifies the device
• If required (as a response to 401/407 message), the gateway sends REGISTER with
authentication

76
Registration Call Flow

SIP UA SIP Registrar

Register

401 Unauthorized

Register

200 OK

77
Registration Call Flow

REGISTER sips:myregistrar.mynet.com SIP/2.0

Via: SIP/2.0/TLS user1.mynet.com:5061; branch=z9hG4bKnashds7
Max-Forwards: 70
From: myname <sips:myname@mynet. com>; tag=a73kszlfl
To: myname <sips:myname@mynet. com>
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]>
Content-Length: 0

SIP/2.0 401 Unauthorized

Via: SIP/2.0/TLS user1.mynet.com:5061; branch=z9hG4bKnashds7
Max-Forwards: 70
From: myname <sips:myname@mynet. com>; tag=a73kszlfl
To: myname <sips:myname@mynet. com>
Call-ID: [email protected]
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“mynet.com", qop="auth",
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale=FALSE, algorithm=MD5
Content-Length: 0

78
Registration Call Flow

REGISTER sips:myregistrar.mynet.com SIP/2.0

Via: SIP/2.0/TLS user1.mynet.com:5061; branch=z9hG4bKnashds7
Max-Forwards: 70
From: myname <sips:myname@mynet. com>; tag=a73kszlfl
To: myname <sips:myname@mynet. com>
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]>
Authorization: Digest username=“myname", realm=“mynet.com" ,
nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri=" sips:myregistrar.mynet.com ",
response="dfe56131d1958046689d83306477ecc"
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/TLS user1.mynet.com:5061; branch=z9hG4bKnashds7; received 15.0.0.1
From: myname <sips:myname@mynet. com>; tag=a73kszlfl
To: myname <sips:myname@mynet. com>;tag=37GkEhwl6
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]> ;expires=3600
Content-Length: 0

79
Enterprise PSTN & Data Network

Headquarters
Branch

PSTN

IP

Telecommuter

80
FXS Gateways
• FXS (Foreign Exchange Station) – Emulates a PSTN/PBX.
Provides battery power, sends dial tones and generates ringing voltage.
A standard telephone / fax machine plugs into such an interface to receive telephone
services.
• FXS gateways convert (in real time) loop start signaling to SIP and variable electric currents to
RTP

FXS Gateway

IP Phone
IP Signaling
IP
Local Loop IP Voice

81
FXO Gateways
• FXO (Foreign Exchange Office) – Generates the on-hook and off-hook indicators used to
signal a loop closure at the FXS's end of the circuit.
Analog telephone handsets, fax machines and (analogue) modems are FXO devices
• FXO gateways convert (in real time) loop start signaling to SIP and variable electric currents
to RTP

FXO Gateway

PBX

IP Signaling
IP Phone
IP Voice
IP Local Loop

82
FXS Call Flow
Analog Phone
FXS Gateway
IP

Off-hook Dial Tone

Dialing
INVITE

100 Trying

180 Ringing
Ringback Tone
200 OK

ACK

Voice

On-hook BYE

200 OK

83
Digital Gateway

• Digital gateways convert (in real time) ISDN or CAS signaling to SIP and PCM to
RTP
PBX

E1 / T1

Mediant 1000 Mediant 2000

IP Signaling
E1 / T1
IP PSTN
IP Voice PCM

84
ISDN Call Flow
PBX Digital Gateway

IP

Setup
INVITE
Call Proceeding
100 Trying

180 Ringing
Alert

200 OK
Connect
ACK

Voice
Disconnect

Release BYE

Release Complete 200 OK

85
Media Processing

Digital Audio Source (PCM)

IP RTP
Network
Digital Gateway Analog Gateway

Encoder Jitter Buffer

Packetization Decoder

RTP

Analog Audio Source

86
Lesson 4

AudioCodes Documentation
Lesson Objectives

• After completing this lesson, you will:

• Understand how to obtain technical documentation from AudioCodes’ Web site

• Be familiar with the different documents that AudioCodes publishes regularly for its'
products
• Understand how to use the documents for configuration and maintenances purposes

88
Obtaining AudioCodes Documentation

• You can access all AudioCodes' documentation from AudioCodes Web site
• This includes:
• Technical documentation (user manuals, hardware installation manuals, configuration
notes and release notes)
• Homologation material (regulatory information)
• Partner/channel material (interoperability guides etc.)
• Marketing material (white papers, application notes, product notices, etc.)

89
Obtaining Document

90
Obtaining Document (Cont.)

• Use the following filters to search for you document:

• Product Family: Choose the family to which the product belongs
• Product: Choose the required product
• Software Version: Choose an option that is displayed in the format Version <version>
(e.g. Version 7.2)
• Documentation Type: Choose the type of document (e.g. User Manuals)

91
Specific Documentation

• Analog Gateways (MediaPack family):

• MP-11x & MP-124, MP-1288
• Digital Gateways and/or SBCs (Mediant family):
• Mediant 500L/500, 800, 1000B, 2600, 3000, 4000, 9000, SW Virtual Edition,
SW Server Edition
• For each product, the following documents are published per release:
• User’s Manual
• Hardware Installation Manual

92
Enterprise Gateways and SBCs User’s Manual

• Main document for configuration and maintenance

• Divided into parts, such as:
• Overview of the product
• Getting started
• Management tools
• General System Settings
• General Configuration
• Specific applications’ description and configuration
• Maintenance
• Status, Performance Monitoring and Reporting
• Diagnostics
• Appendixes

• Identified by software release version

93
Hardware Installation Manual

• Hardware description and step-by-step procedures

for installing and cabling the device
• Divided into chapters, such as:
• Overview of the product
• Unpacking the device
• Physical description
• Mounting the device
• Cabling the device
• Hardware maintenance

94
Additional Documentation

• Besides the previous manuals there are other

useful documents
• Release Notes
• One per software release
• Includes:
• New features
• Updates
• Bugs fixing
• Workarounds on existing constraints
• Others

95
Additional Documentation

• Complementary Guides
• Includes
• Reference Guides
• Design Guides
• Security Guidelines
• Utilities Guides
• Others
• Identified by software release version

96
Additional Documentation

• Configuration Notes
• Document providing a detailed description on how
to configure a specific feature/function/application
for a product
• Normally referenced by the User’s Manual

97
Lesson 5

Media Pack Overview

Lesson Objectives

After completing this lesson, you’ll:

• Understand the difference between FXS and FXO functionality

• Be familiar with MediaPack gateway hardware

99
Typical MediaPack VoIP Application

100
FXS vs. FXO

FXS FXO
Foreign eXchange Subscriber Foreign eXchange Office
Receives hook flash, off hook, on Generates hook flash, off hook, on
hook hook
Receives and Transmits DTMF Receives and Transmits DTMF

Provides Ring and Dial Tone, Caller ID Receives Caller ID, Ring Tone, Dial
Tone

101
Analog Gateways Overview
• Analog FXS and FXO VoIP gateways
• Available configurations:
• MP-112 featuring 2 FXS ports
• MP-114 featuring 4 FXS / FXO / Mixed FXS + FXO ports
• MP-118 featuring 8 FXS / FXO / Mixed FXS + FXO ports
• MP-124 featuring 24 FXS ports
• MP-1288 featuring up to 288 FXS ports
• Firmware file:
• MP-11x gateways (FXS and FXO) use the same firmware (.cmp) file *
• MP-124 gateway requires it own firmware file *
• MP-1288 gateway requires it own firmware file
Note: The latest maintenance firmware version for MP-11x and MP-124 is 6.6
102
Analog Gateways Portfolio

MP-112 MP-114 MP-118 MP-124 MP-1288

Number of
analog ports
2 4 8 24 288

FXS / FXO FXS FXS / FXO FXS / FXO FXS FXS

Power Supply AC AC AC AC / DC AC / DC

103
MediaPack MP-11x
• MP-11x with 2, 4 or 8 FXS/FXO Ports
• RJ-11 connector, RS-232 and Ethernet
• Same Firmware for all MP-11x
MP-11x Front Panel

104
MP-11x Rear Panel

105
MediaPack MP-124
• MP-124 with 24 FXS Ports
• 50-pin Telco Connector, RS-232 and Ethernet
• AC or DC Power Supply
MP-124 Front Panel

106
MP-124 Rear Panel

Item Label Component Description

1 Protective earthing screw
AC power supply socket.
100-240 V~ / 50 - 60Hz 0.8A
Note: Applicable only to the AC-powered model.
2
DC inlet for a DC terminal block.
48V 1.3A
Note: Applicable only to the DC-powered model.
3 ANALOG FXS LINES 1–24 50-pin Telco connector, providing up to 24 analog lines.
4 RS-232 DB-9-pin male port for serial (RS-232) communication.
5 ETHERNET RJ-45 port for 10/100Base-TX Ethernet interface.

107
MP-1288 Overview
• 19” x 3U Chassis
• Single CPU module
• 4 Analog blades, each supporting 72 ports
• 3 x 24 Port Connectors (same 50-pin connector as MP-124)
• 1+1 AC Power Supplies
• Front to Rear Cooling
• Extractable fan tray
• 1+1 Gig ETH connection
• DSPs on each Blade
• Hot-swappable
• Supports short and long haul up to 7.5 Km
• SBC functionality
• 3 lifeline ports per blade
• DSPs on each Blade
• Support for emergency/elevator phones that require higher loop current and increased ring voltage
• Integrated protection against surge damages on FXS ports
• Support line test functionality

108
MP-1288 Overview
MP-1288 Front Panel

Item # Label Description

1 - Fan Tray cover

2 SYS / TEL / PWR / FAN Front-panel LEDs

MP-1288 Rear Panel Item # Label Description

CPU module providing the central
1 CPU processing unit and various
network port interfaces
2 PS1 / PS2 Power Supply modules
Blades: S1 / S2 / S3 / S4
FXS blades providing FXS port
3 FXS Ports: FXS 1-24 / interfaces
FXS 25-48 / FXS 49-72
Protective grounding for
connecting a grounding lug for
4 chassis ground connection for ESD-
preventive equipment or a
grounding wire
109
Analog Lifeline Support

• Provides a wired analog POTS connection to any PSTN or PBX FXS port when power fails or
when the network connection fails
• Available configurations:
• FXS only: A single Lifeline connected to Port #1 using a splitter
• Mixed FXS and FXO: Splitter not required - all FXS ports automatically connected to FXO ports
(e.g., FXS Port 1 to FXO Port 5)
• FXO only: Lifeline not available
• Activated by parameter LifeLineType
Telephone PBX/PSTN

Telephone
PBX/PSTN
110
Analog Lifeline Support – MP1288

• Each FXS blade supports up to three FXS Lifelines, one per FXS connector
• For each connector, the first channel provides the connection to the Lifeline
extension and the last channel is the Lifeline interface providing the connection to
the PSTN / PBX
• For FXS connector labeled FXS 1-24, channel 1 is the Lifeline extension and channel
25 is the Lifeline interface for the PSTN / PBX

Telephone PBX/PSTN

111
Lesson 6

Assigning Networking Parameters

Objectives

• Assigning IP networking parameters

113
Assigning Networking Parameters

• There are five ways to assigned network parameters:

• HTTP using Web browser
• BootP
• DHCP
• RS-232
• Voice Menu (MP-11x FXS only)

114
Default Factory IP Address

Product Default
FXS and FXS / FXO devices – 10.1.10.10
MediaPack MP-11x and MP-124 Analog GW
FXO devices – 10.1.10.11
Software E-SBC
192.168.0.1/24
Mediant 9000 E-SBC
MediaPack MP-1288
Mediant 500 E-SBC
Mediant 800 E-SBC 192.168.0.2/24
Mediant 1000 E-SBC
Mediant 2600/4000 E-SBC
Mediant 500L MSBR LAN Data – 192.168.0.1/24 (DHCP Server enable)
Mediant 500 MSBR LAN Voice – 192.168.0.2/24
Mediant 800 MSBR WAN Data – DHCP Client

115
Assigning IP Address – HTTP

• Connect the gateway to a PC using straight thru Ethernet Cat5e

• If the PC does not support Auto-MDIX then connect
• Via Hub/Switch
• Directly using an Ethernet cross-over cable
• Change the PC’s IP address and subnet mask to correspond with the gateway's
factory default networking parameters
• Open a Web browser and access the Web interface
• Change the networking parameters via ‘IP Settings’
(Configuration tab > VoIP menu > Network sub-menu > IP Settings)
• Reconnect the gateway and your PC (if necessary) to the network
• Restore your PC’s IP address and subnet mask to their original settings
116
Assigning IP Address – HTTP (v6.6)

Use only to define Multiple

IPs and VLANs

117
Assigning IP Address – HTTP (v7.2)

118
Assigning IP Address – BootP

• Bootstrap protocol allows a host to configure itself dynamically

• BootP provides three main services:
• Assigns IP address and networking parameters
• Provides the name of the software (cmp) file and configuration (ini) file to be loaded by
the gateway (via TFTP)
• Provides the IP address of the TFTP server
• MP-11x
• Hardware reset triggers a BootP request
• E-SBC/MSBR
• BootP request on startup is not supported on Mediants E-SBC/MSBR
• To force a BootP request, press the Reset button for 30 seconds
(Rescue Mode – Restore factory default)

119
AudioCodes’ BootP & TFTP Utility – Preferences

All files that are to be

loaded to the gateway
should be placed in the
TFTP directory

120
BootP & TFTP Utility – Client Configuration

MAC address

Enable Device

Assign IP-Address, Subnet, and

Default GW parameters to the GW

IP-Address of TFTP Server

CMP file to load

Burn CMP SW to flash

Ini file to load

121
BootP & TFTP Utility – Monitor

Monitoring
Area

122
Question

• Why does the gateway revert to the old software version after it is reset?
• When loaded via BootP/TFTP, the CMP file must be burned to flash
(using the –fb option)
• Otherwise the CMP file is only stored in the volatile memory (RAM)

123
Assigning IP Address – DHCP

• DHCP – Dynamic Host Control Protocol

• Provides a mechanism for allocating IP addresses dynamically so that addresses can
be reused
• After the gateway is powered up, it attempts to communicate with a BootP server, if
there is no response and if DHCP is enabled (DHCPEnable = 1), the gateway attempts
to obtain its IP address and other network parameters from the DHCP server
• The DHCP server leases one of the available addresses for a specified amount of
time; when it expires, the gateway renews the lease

124
Assigning IP Address – DHCP (v6.6)

• Enable DHCP using WEB GUI

125
Assigning IP Address – DHCP (v7.2)

• Enable DHCP using WEB GUI

126
Question

• The Gateway doesn’t maintain its IP address, what can I do?

• Many DHCP servers are backward-compatible with BootP protocol; these DHCP servers
reply to the bootP requests that are sent by the gateway
• To configure the gateway to ignore these bootP replies, set the parameter:
BootPSelectiveEnable = 1
• This mechanism enables the gateways to accept only BootP replies that contain the text
‘AUDC’ in the vendor specific information field

127
Assigning IP Address – RS-232 MPs (CMD Shell)
• Connect the gateway's RS-232 port to your PC
• Use the following communications port settings:
• Baud Rate: 115, 200 bps
• Data bits: 8
• Parity: None
• Stop bits: 1
• Flow control: None
• At the CLI prompt, type conf and then press <Enter>
• To check the current network parameters, at the prompt type GCP IP, and then press
<Enter>
• Change the network settings by typing:
• SCP IP [ip_address] [subnet_mask] [default_gateway]
e.g., SCP IP 10.13.77.7 255.255.0.0 10.13.0.1
• To save the configuration, type SAR and then press <Enter>; the gateway restarts with the new
network settings
128
Assigning IP Address – RS-232 MP-1288/Mediants CLI

• Connect the device's RS-232 port to your PC

• Use these communications port settings:
• Baud Rate: 115200 bps
• Data bits: 8
• Parity: None
• Stop bits: 1
• Flow control: None
• At the CLI prompt, type the username (case sensitive):
• Username: Admin
• At the prompt, type the password (case sensitive):
• Password: Admin
• At the prompt, type the following:
• enable
• At the prompt, type the password again:
• Password: Admin
129
Assigning IP Address – RS-232 MP-1288/Mediants CLI

• Access the network configuration mode:

• # configure network
• Access the Interface table:
• (config-voip)# interface network-if 0
• Configure the IP address:
• (network-if-0)# ip-address <IP address>
• Configure the prefix length:
• (network-if-0)# prefix-length <prefix length / subnet mask>
• Configure the Default Gateway address:
• (network-if-0)# gateway <IP address>
• Exit the Interface table:
• (network-if-0)# exit
• Save the Configuration:
• # write
130
Assigning IP Address – RS-232 (cont.)

Username: Admin
Password: *****

Mediant 800> enable

Password: *****

Mediant 800# configure network

Mediant 800(config-network)# interface network-if 0

Mediant 800(network-if-0)# ip-address 10.15.17.55

Note: Changes to this parameter will take effect when applying the 'activate' or 'exit’ command

Mediant 800(network-if-0)# prefix-length 16

Note: Changes to this parameter will take effect when applying the 'activate' or 'exit' command

Mediant 800(network-if-0)# gateway 10.15.0.1

Note: Changes to this parameter will take effect when applying the 'activate' or 'exit' command

Mediant 800(network-if-0)# exit

Mediant 800(network-if-0)# write

Mediant 800(config-network)# exit

Mediant 800#
After ‘exit’ the address changed. Logon again using the new IP address

131
Assigning IP Address – Voice Menu

• Applies only to MP-11x FXS only

• Connect a telephone to one of the FXS ports and dial ***12345 (three stars
followed by the digits 1, 2, 3, 4, 5)
• Wait for the 'configuration menu' voice prompt to be played and follow the
instructions
• The following configuration parameters can be queried or modified via the voice
menu:
1. IP Address
2. Subnet Mask
3. Default gateway IP address
4. Primary DNS server IP address
5. DHCP enable / disable
132
Question

• I forgot the IP address of the Gateway, what can I do?

• The easiest method to locate the IP address of the gateway is to open the Wireshark
application, reset the gateway and wait for it to startup
• When the gateway starts up, it sends a Gratuitous ARP (GARP) message with its IP
address

133
Lesson 7
Management & Maintenance
Web Interface - MediaPack MP-11X/MP-124
Lesson Objectives

• After completing this lesson, you will be able to:

• Log in to the web interface
• Identify the components of the web interface
• Identify the navigation features of the web interface
• Use the search feature of the web interface
• Use the scenario feature of the web interface

135
Management and Maintenance Interfaces

Embedded Web Server Command Line Interface (CLI)

(only for maintenance and debugging)

Configuration File
(referred to as the ini file) REST-based programs
(such as AudioCodes’ OVOC)

136
Accessing the Web Interface

• Default Username: Admin

• Default Password: Admin

• Restore Gateway Username / Password to defaults:

• If your configuration is backed up as an ini file, add parameter ResetWebPassword=1 to the ini
file and load it via BootP
• For MSBG, press the Reset button for 6 seconds to restore the configuration to default, and then
load the backed-up ini file

137
Getting Acquainted with the Web Interface

Title Bar Tool Bar

Navigatio
n
Bar

Navigatio
n
Tree

Note: Includes GW/SBCs up to release 7.0 Work

pane

138
Getting Acquainted with the Web Interface
Title Bar Tool Bar

Navigation
Bar

Navigation
Tree

Work pane

Note: Same Interface Includes GW/SBCs up to release 7.0

139
Getting Acquainted with the Web (cont’d)

• Title Bar
• Displays the corporate logo and product name
• Tool Bar
• Provides frequently required command buttons for configuration
• Navigation Bar
• Provides tabs for accessing the configuration menus, creating a scenario, and searching
parameters
• Navigation Tree
• Displays the elements pertaining to the tab selected on the Navigation Bar
• Work pane
• Displays configuration pages where all configuration is performed

140
Title Bar

• Title Bar
• Displays the corporate logo and product name
• Company logo
• Loaded image file, gif, png or jpeg
• Loaded text via ini file
• Device Name
Company Device
logo Name

141
Tool Bar

• Tool Bar

• Provides frequently required command buttons for configuration

142
Tool Bar (cont'd)

• If you click the Submit button after modifying parameters that take effect only after
a device reset, the Tool Bar displays ‘Reset’ (in red)
• This is a reminder to later save ('burn') your settings to flash memory and reset the
device

143
Navigation Bar

• Navigation Bar
• Provides tabs for accessing the configuration menus, creating a scenario, and searching
parameters

Icon Description
Includes all gateway parameters and tables

Allows you to perform software update, Upgrade Feature Key, and

maintenance actions

Allows you to monitor and view gateway information & statistics

Allows you to create your own ‘menu’ with pages selected from the
menus in the Navigation Tree. Not supported anymore in version 6.8
Search engine enabling searching for any ini file parameter that is
configurable in the Web interface

144
Navigation Tree
Basic / Full
• Navigation Tree view

• Displays the elements pertaining to the tab

selected on the Navigation Bar
• Expand or collapse the Navigation Tree:
Show / Hide
• Basic displays only commonly-used menus button

• Advanced displays all menus (in releases

previous to 6.8 it was known as Full)
• Hide the Navigation pane to provide more
space for elements displayed in the Work
pane

145
Work Pane

• Work pane
• Displays configuration pages where all configuration is performed
• Displaying Basic or Advanced Parameter List
• Advanced Parameter List displays all parameters
• Basic Parameter List displays only common parameters
• When the Navigation Tree is in ‘Advanced' mode, the 'Advanced Parameter List'
view is displayed Advanced/Basic
Parameter List

146
Work Pane (cont'd)

• Some pages provide groups of parameters which can be hidden/shown by clicking

the group name button that appears above each group

147
Modifying/Saving Parameters

• When changing parameter values, the Edit symbol appears

• To save configuration changes to volatile memory (RAM), click the Submit button
• Modifications to parameters with on-the-fly capabilities are immediately applied to
the device and immediately take effect
• Parameters displayed with a lightning symbol are not changeable on the fly and
require a gateway reset

148
Modified Parameter

• In case of invalid parameter value, after the select Submit, you will get a Red Alert

149
Modifying/Saving Parameters (cont'd)

• Parameters saved to volatile memory (by clicking Submit ) revert to their

previous settings after a hardware or software reset (or if the device is powered
down)
• To retain parameter changes (whether performed on-the-fly or not), save (‘Burn')
them to the device's flash memory

150
Maintenance Actions

• Maintenance tab > Maintenance menu > Maintenance Actions

151
Maintenance Actions

• Reset: After a Web reset, the gateway starts from Flash and doesn’t issue BootP
requests
• LOCK: The gateway doesn't accept any new incoming calls
• BURN: Save the running configuration to the memory
• Graceful: Shutdown will perform only after X time or no more active traffic exists

152
Navigation Bar – Configuration

• Contain the parameters for configuring the Gateway

153
Navigation Bar – Management

• Management related menu:

• Maintenance Actions (reset, lock/unlock, save configuration)
• Loading auxiliary files, (ini, CAS, Dial plan, Call progress tones,
pre-recorder tones, user information)
• Software upgrade key
• Allows you to load a new Software Upgrade Key to the device
• You can upgrade or change the device supported items by
purchasing a new Software Upgrade Key to match your
requirements
• Software upgrade wizard
• Allows you to upgrade the device firmware (.cmp file)
as well as load an ini file and/or auxiliary files
(such as Call Progress Tones)
• Configuration file (backup and restore)
154
Navigation Bar – Status & Diagnostics
• Contains Gateway operating status and diagnostics related information

155
Navigation Bar – Search

• The search engine enables searching for any ini file parameter that is configurable
in the Web interface

Search
Field Parameter
highlighted
in page

Searched
Results

156
Admin Page

• Use AdminPage to configure parameters that don’t appear in the regular Web
menu

Case
Sensitive

157
ini File
Configuration ini File

• The gateway can be configured by loading the ini file

• The ini file can be loaded via BootP/TFTP, Web interface, or using the automatic
update mechanism
• When a parameter is missing from the ini file, its default is assigned
• To restore the defaults, use an empty ini file (except for the incremental option via
the Auxiliary Files page)

159
Configuration File Example

Serial Number = Decimal representation of the last

6 digits of the MAC address (i.e., 00:90:8F:1A:F8:6A)

6.60 – Major software version

A – Indicates that this is a SIP version (e.g., not MGCP)
347.002 – Minor software version

160
Configuration File (ini file)

161
ini File Parameters
• The ini file can be loaded via BootP/TFTP, Web interface, or using the automatic update mechanism
• Case insensitive
• Lines beginning with semi-colon (;) as first character are ignored
• Carriage Return must be each line’s final character
• Number of spaces before and after equal ( = ) is irrelevant
• Values of string parameters must be placed between two single quotes ( ‘ ’ )
• Syntax errors in value can cause unexpected errors (may be set to wrong values)
• Syntax error in the parameter name is ignored (error message is generated)
• When a parameter is missing from the ini file, its default is assigned
• Subsection names are optional
[Sub Section Name]
Parameter_Name = Parameter_Value
Parameter_Name = Parameter_Value

; REMARK
162
ini File Table Parameters

• Tables are used in ini files to represent parameters that have several instances
(e.g., Coders, Proxy servers, Routing tables, etc.)
• Examples:

163
AudioCodes INI Viewer & Editor
• A simple viewer and editor for configuration (INI) files used by AudioCodes Media Gateway
and Session Border Controller (SBC) products
• Two Modes:
• View Mode: View Mode
• Standalone and Table Edit Mode
parameters can be viewed in
a very friendly way
• Edit Mode:
• Standalone and Table
parameters can be edited
(modified, added, removed,
etc.) for a very easy way of
changing their contents
• Once this is done, the new
INI file can be saved and
uploaded to the device in
order to apply the new
configuration

164
AudioCodes INI Viewer & Editor

• Table Parameters in View Mode

165
AudioCodes INI Viewer & Editor

• Table Parameters in Edit Mode

166
Restore and Back Up the ini File
• Device Actions > Load/Save Configuration File
• Maintenance tab > Software Update menu > Configuration File

To Back Up the ini file

To Restore the ini file

167
Loading Incremental ini File

• Maintenance tab > Software Update menu > Load Auxiliary Files
• Used to load specific parameters to the gateway
• Parameters not included in the ini file are retained (and not reset to defaults)
• After loading the ini file the gateway doesn’t reset automatically

168
Upgrading & Downgrading Software

• The gateway can be updated with software (cmp file), configuration (ini file),
auxiliary files and license key (not relevant to MPs) using:
• Web interface
• BootP/TFTP utility
• Automatic Update Mechanism

169
Upgrading/Downgrading Software – via Web Interface
• Maintenance tab > Software Update menu > Software Upgrade Wizard

170
Upgrading/Downgrading Software – via Web Interface

171
Restoring Factory Default Settings
• The device's factory defaults can be restored using :

• Reset button

• CLI
• CONFiguration > RestoreFactorySettings

172
Lesson 8
Management & Maintenance
Web Interface - MediaPack MP-1288/Mediants
Accessing the Web Interface

Default Username: Admin

Default Password: Admin
174
GUI Areas

Toolbar providing
Company Logo Menu Bar Containing the Menus frequently required
• Setup command buttons
• Monitor
• Troubleshoot

Alarm bell icon: Displays the

number of active alarms
generated by the device

Button displaying
the username of
the currently
logged in user

175
GUI Areas

Search box for

Tab bar containing tabs pertaining to the selected menu: searching parameter
names and values
• Setup menu:
• IP Network
• Signaling & Media
• Administration
• Monitor menu:
• Monitor
• Troubleshoot menu:
• Troubleshoot

176
GUI Areas
Back and Forward buttons that enable quick-
and-easy navigation through previously
opened pages

SRD filter
When your configuration includes multiple SRDs, you
can filter tables in the Web interface by a specific SRD

177
GUI Areas

Work pane:
Where configuration pages are displayed

178
Tool Bar

Button Description
Save Saves parameter settings to flash memory
Reset Resets the device
Opens a drop-down menu list with frequently needed commands:
Configuration Files to load or save an ini file
Auxiliary File to load auxiliary files such as: Dial Plans, Call Progress Tones, others
Actions
License Key to determine features, capabilities and available resources
Software Upgrade to upgrade the device's software
Configuration wizard

Displays the number of active alarms generated by the device

Opens a drop-down menu and:

Logon Name Shows the logged in user’s access level and session time
(i.e. Admin) Allow password change
Allows to Logout
179
Modifying/Saving Parameters

• When changing parameter values, the changed parameter has a yellow background
• To save configuration changes to volatile memory (RAM), click the Apply button

• Modifications to parameters with on-the-fly capabilities are immediately applied to

the device and immediately take effect
• Parameters displayed with a lightning symbol are not changeable on-the-fly and
require a device reset

180
Modifying/Saving Parameters

• If you click the Apply button after modifying parameters a red rectangle appears
surrounding the Save button
• This is a reminder to save your settings to flash memory

• If you click the Apply button after modifying parameters that take effect only after a
device reset, a red rectangle appears surrounding the both, the Save and Reset
buttons
• This is a reminder to later save your settings to flash memory and reset the device

181
Stand-alone Parameters
• Parameters that are not contained in a table are referred to as stand-alone parameters

Stand-alone parameters

182
Stand-alone Parameters Configuration
• Parameters not requiring a device reset

3. Click Save
(Changes are saved to the
non-volatile memory (flash))

1. Modify the parameter's

value as desired

4. Click Yes

2. Click Apply
(Changes are saved to the volatile memory (RAM))

183
Stand-alone Parameters Configuration
• Parameters requiring a device reset

3. Click Save
(Changes are saved to the
non-volatile memory (flash))
4. Click Reset
(the Maintenance
Actions page opens)

1. Modify the parameter's

value as desired

2. Click Apply
(Changes are saved to the volatile
memory (RAM))

184
Stand-alone Parameters Configuration
• Resetting the device

2. Click OK

Please note
1. Click Reset
(the device saves the changes to flash memory and then resets)
185
Stand-alone Parameters Configuration

• Restarting the devices

186
Stand-alone Parameters Indications Meaning

Parameters changed and not applied are highlighted

A dot appears next to parameters changed from their

default values and when the Apply button was clicked

Changes on parameters displaying a lightning-bolt icon,

require to be saved to flash memory followed by a device
reset for your changes to take effect

Typically required parameters are displayed in bold font

An invalid value for a parameter reverts to its previous

value and is surrounded by a colored border

To get help on a parameter, hover your mouse over the

parameter's field
A pop-up help appears, displaying a brief description of
the parameter

187
Table Parameters – General Description

Page title (name of table) Navigation bar for scrolling Search tool for searching
Also displays the number of through the table's pages parameters and values
configured rows as well as the Sort can be done
number of invalid rows by any column

Added table rows displaying

Adds a new row to the table only some of the table
Modifies the selected row parameters
Deletes the selected row

Detailed view of a selected row, displaying all parameters

Link to open the "child" table of the "parent" table.

Only appears if the table has a "child" table
188
Table Syntax
• The table is divided into two main areas: Matching characteristics and Action to take
• If the incoming call matches the characteristics of a rule, then the call is sent to the destination
configured for that rule
• Non-configured parameter fields
may appear with different values,
for example, “-1”, “0” or empty

189
Fields to Match

• Device attempts to match patterns at the top of the table first (first match)
• More specific rules should be at the top and more generic ones at the bottom

Take the rule up

‘551’ will never match because ’55’ matches

every prefix that starts with ’55’

190
Numbers Notation for Routing and Manipulation
• Flexible numbers notations for describing the prefix and/or suffix source and/or destination
phone numbers and SIP URI user names:

▪ Prefix [n-m] or Suffix (n-m)

▪ Represents a range of numbers

▪ Prefix [n,m,...] or Suffix (n,m,...) Destination Phone Prefix Source Phone Prefix
▪ Represents multiple numbers 1 9x*
▪ Multiple ranges such as [n-m,s-t] are also supported 2[2,6,7,9] 1xxx
▪ Up to three digits can be used to denote each number 2[1-4,7,9] 1xxx#
[100-150,222,244,300-499] 1*
▪ x (letter ‘x’) 6[100-300] (99)
▪ Represents any single digit 976(99) 2[1-4]
6[100-300]# *
▪ # (Pound symbol)
▪ Represents the end of a number
* *

▪ * (asterisk symbol)
▪ Represents any number

191
Numbers Notation
• Examples:
• [5200-5300]#
• represents all numbers from 5200 to 5300
• [2,3,4]xxx#
• represents four-digit numbers that start with 2, 3 or 4 (2000-4999)
• 54324
• represents any number that starts with 54324
• 54324xx#
• represents seven-digit numbers that start with 54324
• 123[100-200]#
• represents six-digit numbers that start with 123 (123100 to 123200)
• (100)
• represents any number that finishes with 100
• (266[1-9])
• represents any number that finishes with 2661 to 2669

192
Assigning Rows from other Tables

• Tables may contain parameters assigned a value which is a row referenced from
another table

A View button opens the

row-referenced table

193
Assigning Rows from other Tables

• For example, after pressing the View button pointing to the Network Interface,
the referenced table web page is opened

A View button opens the

row-referenced table

194
Table Parameters Invalid Values Indications
• When adding a row:
• If a mandatory parameter’s value, which is a row referenced from another table is not assigned,
after clicking Apply, an error message is displayed at the bottom of the dialog box
• Clicking Cancel closes the dialog box and the row is not added to the table
• To add the row, you must configure the parameter

195
Table Parameters Invalid Values Indications
• When editing a row:
• If a parameter’s configuration is changed so that it's no longer assigned with a referenced
row from another table, when the dialog box is closed, the Invalid Line icon appears for
the table in which the parameter is configured, in the shown locations:

3. Item in the Navigation tree 1. Page title of the table. The total number of invalid rows in the
that opens the table table is also displayed with the icon

2. 'Index' column of the row to which the parameter belongs

196
Table Parameters Invalid Values Indications

• When a parameter assigned a value which is an invalid row referenced from

another
• The Invalid Reference Line Icon is displayed for the table in which the parameter is
configured, in the shown locations

1. Page title of the table. The total number of invalid rows in the
table is also displayed with the icon

2. 'Index' column of the row to which the parameter belongs

3. Item in the Navigation tree that opens the table

197
Searching for Configuration Parameters

• Parameter names (standalone or table) and values can be searched in the Web
interface
• The search key can include the full parameter name (Web or ini file name) or a substring
of it
• For a substring, all parameters containing the substring in their names are listed in the
search result
• The search key for a parameter value can include alphanumeric and certain characters
• The key can be a complete value or a partial value
• When the device completes the search, it displays a list of found results based on
the search key
• Each possible result, when clicked, opens the page on which the parameter or value is
located

198
Searching for Configuration Parameters

Search can
be by name
or by value

199
Setup Menu

• 3 Options:

• IP Network

• Signaling & Media

• Administration

200
Setup Menu: IP Network Option
• Home Page: NETWORK VIEW
• Shows a graphical display of the core networking entities
• IP interfaces
• VLANs (Ethernet Devices)
• Ethernet Groups
• Physical Ethernet ports
• Enables the administrator to easily build and view the main network topology
• Other Pages
• Networking Core Entities
• Security
• Quality
• DNS
• WEB Services
• HTTP Proxy
• Radius & LDAP
• Advanced
201
Setup Menu: IP Network Option
• Home Page: NETWORK VIEW

IP Interfaces can be added, VLANs can be

edited, viewed or deleted added, edited,
viewed or deleted

Ethernet Groups
can be, edited
or viewed

Physical Ports
can be, edited
or viewed

202
Setup Menu: Signaling & Media Option
• Home Page: TOPOLOGY VIEW
• Shows a graphical display of the core SIP configuration entities
• IP Groups
• SIP Interfaces
• Media Realms
• Enables the administrator to easily build and view the SIP topology
• Other Pages
• Signaling and Media Core Entities
• Gateway
• Media
• Coders and Profiles
• SBC
• SIP Definition
• Message Manipulation
• Intrusion Detection
• SIP Recording

203
Setup Menu: Signaling & Media Option
• Home Page: TOPOLOGY VIEW

IP Groups can
be added

Trunk Groups
Tel view (i.e. related can be added IP top view (i.e.
to the PSTN) related to the WAN)

SIP Interfaces can be SIP Interfaces can be Media Realms can be

added and shown at added and shown at added and shown at
the top or bottom the top or bottom the top or bottom
(GW application) (SBC application)

The links between SIP

Interfaces, Media Realms
and IP Groups are shown IP bottom view (i.e.
related to the LAN)

IP Groups can
be added
204
Setup Menu: Signaling & Media Option
• Home Page: TOPOLOGY VIEW

Hover to see the

basic configuration

Click to edit, show,

or delete
parameters or tables

Hover to see
the basic
configuration

Hover to see the

basic configuration

Click to edit, show,

or delete
parameters or tables Click to edit, show,
or delete
parameters or tables

205
Setup Menu: Signaling & Media Option
• Home Page: TOPOLOGY VIEW

Direct links to the

Direct links to SBC’s main
the Gateway’s parameters and
main parameters tables
and tables

Indications of valid or invalid configuration on tables or parameters

206
Setup Menu: Administration Option

• Home Page: TIME & DATE

• Shows and allows to configure parameters related to:
• Local Time
• NTP Information
• Time Zone
• Other Pages
• WEB & CLI pages
• SNMP Pages
• Maintenance

207
Setup Menu: Administration Option
• Home Page: TIME & DATE

Displays and allows Displays and allows

to configure the local to configure the
time and date UTC, offset and DST

Displays and allows

to configure the
NTP server
information

208
Web Local Users Table

Username & Password

User levels:
• Monitor
• Administrator
• Security Administrator
• Master

209
Maintenance Actions
• Reset Device: After a Web reset, the device starts from Flash
• Lock: The device doesn't accept any new incoming calls
• Save to Flash: Save the running configuration to the memory
• Graceful Option: Shutdown will perform only after X configured sec. or no more active traffic
exists

210
Maintenance: Configuration File

To restore the defaults, use an empty ini file (except

for the incremental option via the Auxiliary Files page –
later more on this) or ‘Restore Defaults’ with checked
‘Preserve Network and users Configuration’ (option
supported only on Mediant Family Devices)

Configuration, Auxiliary and Certificate files can be

loaded to and saved from the device as a single,
packaged file. The feature is typically used for backup
and loading the backup to other devices.
211
Configuration package files

• Ini.ini
• LOGO.dat
• FAVICON.dat
• CPT.dat
• PRT.dat
• AMD.dat
• SBC_Wizard.dat
• CAS.dat
• DPLN.dat (Dial Plan)
• Certificate files
• DialPlanRule.csv (import only - can load any CSV file. For example, User-Info Table)
212
Maintenance: Auxiliary Files

Various auxiliary files can

be loaded to the device

213
Maintenance: Upgrading & Downgrading Software

• The device can be updated with software (cmp file), configuration (ini file), auxiliary
files and license key using:
• Web interface
• BootP/TFTP utility
• Automatic Update Mechanism

214
Maintenance: License Key

• Supplied with digital gateways (not relevant for MP-1xx)

• Determines features, capabilities and available resources
• Provided in string format or in a txt file to be loaded to the device
• Stored in the device's non-volatile flash memory
• After loading the new key, the device must be reset

215
Maintenance: License Key

216
Maintenance: HA

217
Maintenance: Configuration Wizard
• The SBC configuration wizard provides fast SBC configuration
• Based on a large set of tested interoperability configurations
• User selects a PBX type and service provider SIP trunk type from a list of over 30 PBX
models and 80 SIP trunks
• Data base updates automatically with new PBX models and SIP trunks from the cloud
• Available in both standalone windows app and embedded on the SBC web GUI

218
Monitor Menu

• One Option: Monitor

• Home Page: MONITOR
• Shows a graphical display of the Device
• Device Information
• Alarms Status
• Activity Log
• Enables the administrator to easily view the device’s main information and statuses
• Other Pages
• Performance Monitoring
• VoIP Status
• PSTN Status
• Network Status

219
Monitor Menu
• Home Page: MONITOR

Shows the IP Address, Firmware, Type of Devices and Serial Number

Displays status and

information on the hardware

Displays SBC’s statistics and

information on calls,
transactions and registration

220
Device Information

221
Troubleshoot Menu

• One Option: TROUBLESHOOT

• Home Page: Message Log
• If logging is active, it shows the device’s activity
• Other Pages
• Logging configuration
• Call Detail Record
• Test Calls
• Debug

222
Troubleshoot Menu

• Home Page: MESSAGE LOG

223
AdminPage

• Used to configure parameters that don’t appear in the Web interface

224
Lesson 9

MP-11X & MP-124 - Basic SIP Configuration

Lesson Objectives

After completing this lesson, you will be able to:

• Configure MP-11x for basic calls scenario

226
Endpoint Phone Number Table

• Allows activation of the Gateway ports (Channels)

• The number of endpoints depends on the MP model
• Allows entry of the channels in groups (n-m) or a separate channel number for each line
• The Phone Number value can include up to 50 characters

227
Endpoint Phone Number Table – Hunt Group

• Up to 100 Hunt Group ID’s (0-99)

• If empty, the Gateway uses the default Hunt Group ID (0)
• Once you have defined a Hunt Group, you must configure the parameters in the 'Inbound
IP Routing Table‘ (IP2Tel)

228
Hunt Group Setting
• Allows to configure settings of up to 24 Hunt Groups
• Allows you to select the method for which IP-to-Tel calls are assigned to channels within each Hunt Group
• If no method is selected for a specific Hunt Group, the setting of the global parameter, Channel Select
Mode (SIP General Parameters screen) takes effect

229
Channel Select Mode
• By Dest Phone Number: Selects the port according to the called number
• Cyclic Ascending: Always selects the next higher channel number in the Hunt Group; when the gateway reaches the
highest channel number, it selects the lowest channel number and then starts ascending again
• Ascending: Always starts at the lowest channel number; if this channel is not available, it selects the next highest one
• Cyclic Descending
• Descending
• Dest Number + Cyclic Ascending: First selects the port according to the called number, if the called number isn't found,
then it selects the next available channel in ascending cyclic order. If the called number is found and the port associated
with this number is busy, the call is released
• By Source Phone Number: Selects the port according to the calling number
• Ring to Hunt Group: The device allocates IP-to-Tel calls to all the FXS ports in the Hunt Group. When a call is received
for the Hunt Group, all telephones connected to the FXS ports belonging to the Hunt Group start ringing. The call is
received by the first telephone that answers the call (after which the other phones stop ringing). This option is
applicable only to FXS interfaces
• Dest Number + Ascending: First selects the port according to the called number, If the number is not located or the
channel is unavailable (e.g., busy), the device searches in ascending order for the next available channel in the Hunt
Group
• Note: If this parameter is not configured for the Hunt Group, then its channel select method is according to the global
parameter, ChannelSelectMode.
230
Q: How do I reduce glare symptoms?

• Glare occurs when both ends of a telephone

line or trunk are seized at the same time PSTN

(user tries to make a call and simultaneously

there is an incoming call)
• Configure the channel select mode to invert
SD

Bay Networks Centillion 1400

P*8x50 ETHER LINK RS 232C INS ACT ALM

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6

PC CARD

the implementation in the PBX e.g. if you

ALM

PBX
PWR ALM FAN0 FAN1 PWR0 PWR1

configure ‘Ascending’ on the gateway side,

then on the PBX side, it is recommended to MP
FXO
configure ‘Descending’
IPIP

231
Coder Table

• Allows you to configure up to 10 coders for the Gateway

• The first coder in the list has the highest priority
• A coder can appear only once in the table
• The Packetization Time determines how many coder payloads are combined into a
single RTP packet
• The Gateway always uses the packetization time requested by the remote side for
sending RTP packets
• Enable/Disable the Silence Suppression option per coder

232
Coder Table

233
IP to Tel (Hunt) Routing Table

• Allows to configure up to 24 inbound call

routing rules
• Routing incoming IP calls to Hunt Groups
• The route based on the matching IP to Tel

characteristics of the incoming call and the IP to Tel

Destination Hunt Group

• In case the Hunt Group is not configured,
the IP to Hunt Group Routing table should
be empty IP to Tel

IP to Tel

234
IP to Tel (Hunt) Routing Table

235
Tel to IP

Two methods
1. Tel to IP Routing Table
• Allows you to configure up to 50 Tel-to-IP
Tel to IP
call routing rules
Tel to IP
• Used to route Tel calls to IP Addresses
when the Proxy isn’t used
• The ‘Destination IP Address’ can be:
• IP Address Tel to IP

• FQDN (Fully Qualified Domain Name) Proxy to

Destination

GW to Proxy
2. Using Default Proxy Tel to IP

236
Tel to IP Routing Table

237
Using Proxy

1. Enable “Use
Default Proxy”

2. Press the
arrow button

3. Configure the Proxy

IP Addresses or FQDN

238
Combine Proxy and Routing Table
• Enable Fallback to Routing Table:
• Determines whether the Gateway falls back to the 'Outbound IP Routing Table' (Tel to IP) for call routing when
Proxy servers are unavailable
• Prefer Routing Table
• Determines whether the Gateway internal routing table takes precedence over a Proxy for routing calls

239
Proxy Name

• Used as Request-URI in REGISTER, INVITE and other SIP Messages

• If not specified, the Proxy IP Address is used instead

INVITE sip:1234@GW1;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.15.8.2;branch=z9hG4bKac547398856
Max-Forwards: 70
From: <sip:[email protected]>;tag=1c547394196
To: <sip:1234@GW1;user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.6.20A.055.003
Content-Type: application/sdp
Content-Length: 314
240
Proxy Name

• Used as Request-URI in REGISTER, INVITE and other SIP Messages

• If not specified, the Proxy IP Address is used instead

INVITE sip:1234@GW1;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.15.8.2;branch=z9hG4bKac547398856
Max-Forwards: 70
From: <sip:[email protected]>;tag=1c547394196
To: <sip:1234@GW1;user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.6.20A.055.003
Content-Type: application/sdp
Content-Length: 314

241
Gateway Name

• Used as the host part of the SIP URI in the From header
• If not specified, Gateway IP Address is used instead
INVITE sip:1234@GW1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.15.8.2;branch=z9hG4bKac13248121
Max-Forwards: 70
From: <sip:[email protected]>;tag=1c13240668
To: <sip:1234@GW1;user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.6.20A.055.003
Content-Type: application/sdp
Content-Length: 312

242
Registration
• Enable Registration: Enables the gateway to register to Proxy/Registrar server
• Registrar Name: If specified, the name is used as the Request-URI in REGISTER messages
• Registrar IP Address: The IP address (or FQDN) and port number (optional) of the Registrar
server
---- Outgoing SIP Message to 10.15.1.1:5060 ----
REGISTER sip:User10 SIP/2.0
Via: SIP/2.0/UDP 10.15.8.2;branch=z9hG4bKac1198896178
Max-Forwards: 70
From: <sip:[email protected]>;tag=1c1198887385
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 4 REGISTER
Contact: <sip:[email protected]:5060>;expires=180
Supported: path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS/v.6.20A.055.003
Content-Length: 0

243
Registration

• Registration Mode: Determines the Gateway registration and authentication

method
• Per Endpoint: Registration and authentication is performed separately for each endpoint
• Per Gateway: Single registration and authentication for the entire Gateway
• Per FXS: Registration and authentication for FXS endpoints
• User name and Password used for registration and authentication with a
Proxy/Registrar server

244
Registration

245
Authentication Table

• Defines a user name and password for authenticating each Gateway port
• Authentication is typically used for FXS interfaces, but can also be used for FXO interfaces
• For configuring whether authentication is done per port or for the entire device,
use the parameter Authentication Mode.
• If authentication is configured for the entire device, the configuration in this table is
ignored
• If the user name or password is not configured in this table, the port's phone
number (configured in the Endpoint Phone Number table and global password
(configured by the global parameter, Password) are used instead for authentication
of the port

246
Authentication Table

247
Maximum Digits In Phone Number
• Defines the maximum number of collected destination number digits that can be received
(i.e., dialed) from the Tel side
• When the Gateway has collected the maximum digits, it’s stops collecting and sends the call
to the destination

248
General Parameters
• Channel Select Mode: The default method for allocating incoming IP-to-Tel calls to Hunt Group
• SIP Transport Type: The default transport layer for outgoing SIP calls (UDP, TCP, or TLS)
• SIP Local Port: The local listening port for SIP messages
• SIP Destination Port: SIP destination port for sending initial SIP requests

249
Registration Status

• Displays whether the Gateway, its endpoints, SIP Accounts, and BRI endpoints are
registered to a SIP Registrar/Proxy server

250
Lesson 10

Basic Debugging Tools

Troubleshooting Guidelines

• Understanding the problem

• What are the expected results?

• What are the actual results?

• Collecting data
• Use the relevant data collection tools for problem investigation

252
Collecting Data

• When reporting a problem, provide AudioCodes Support with:

• Accurate, clear and detailed problem description
• Test setup (network diagram, call direction, etc.)
• Uploaded ini file
• Syslog trace (without missing messages)
• Unfiltered Wireshark
• Advanced (per request):
• PSTN traces for PSTN problems
• DSP traces for problems related to voice quality, Modem/Fax, DTMF detection, etc.

253
What is Syslog?

• Standard for forwarding log messages in an IP network

• A Syslog server is used to remotely record logging information
• Syslog information sent by the gateway is a collection of error, warning and system
messages that record every internal operation of the gateway
• Syslog messages are marked with a sequential number
• A Syslog server usually adds the time the message was received and the source IP
address

254
Syslog Message Format - Example
08:59:10.239 10.15.11.1 local0.notice [S=1974] [SID=a929c9:21:24] ( lgr_sbc)( 1773) Classification Succeeded - Source IP Group #2 (ITSP), - Dest Routing Policy #0
08:59:10.239 10.15.11.1 local0.notice [S=1975] [SID=a929c9:21:24] ( lgr_flow)( 1774) (#3091)SBCRoutesIterator::Change State From: InitialCSRRouting To : InitialRouting
08:59:10.240 10.15.11.1 local0.notice [S=1976] [SID=a929c9:21:24] ( lgr_flow)( 1775) (#3091)SBCRoutesIterator::Change State From: InitialRouting To : AlternativeRouting
08:59:10.241 10.15.11.1 syslog.error 4 packets missing
08:59:10.241 10.15.11.1 local0.notice [S=1981] [SID=a929c9:21:24] ( media_service)( 1780) ServicesMngr: Allocate SBC leg. current active: 1 and max is: 120
08:59:10.242 10.15.11.1 local0.notice [S=1982] [SID=a929c9:21:24] ( lgr_flow)( 1781) (#3091)SBCRoutesIterator::Next route found: Rule #1, Route by: IPGroup , IP Group ID: 1 (SfB), Live:True
08:59:10.242 10.15.11.1 local0.notice [S=1983] [SID=a929c9:21:24] ( lgr_sbc)( 1782) Routing Succeeded -IP2IPRouting Rule #1

Timestamp and Message Sequence Number

IP Address In this example 4 messages
were lost

Type of Message Unique SIP call session and device identifier, SID =
<last 6 characters of device's MAC address>
<number of times device has reset>
<unique SID counter indicating the call session (increments consecutively for each new session; resets to 1 after a device reset)
SID=47ecef:94:69

255
Syslog Types of Messages

• Syslog generates the following types of messages:

• ERROR: Indicates that a problem has been identified that requires immediate handling

• WARNING: Indicates an error that might occur if measures are not taken to prevent it

• NOTICE: Indicates that an unusual event has occurred

• INFO: Indicates an operational message

• DEBUG: Messages used for debugging

256
Enabling Syslog – (v6.6)
• Enable Syslog
• Set Syslog Server IP address and port
• Select the Debug level (recommended ‘Detailed’)

ini parameters

257
Enabling Syslog – (v7.2)
• Enable Syslog
• Set Syslog Server IP address and port
• Select the Syslog level (recommended ‘Detailed’)

258
Message Log

• View the Syslog messages sent by the device

259
AudioCodes Syslog Viewer

• A Syslog application provided with the student utilities kit

Clear On-Line
Syslog

Open/Save Freeze
file Display

Pause/Resume
Syslog

260
AudioCodes Syslog Viewer
• Syslog can be enabled simultaneously in several devices, reporting to the same Syslog Server

Syslog form different IP Addresses can be viewed

261
AudioCodes Syslog Viewer
• SIP/SDP messages are properly arranged to be easily identified for analysis

262
AudioCodes Syslog Viewer
• The SIP/SDP flow diagram can be viewed

SIP Flow
Diagram

263
AudioCodes Syslog Viewer
• Each arrow on the SIP/SDP flow diagram points to the right place in the trace

Highlighted

SIP Flow
Diagram

Points to

264
AudioCodes Syslog Viewer
• CDR info

265
AudioCodes Syslog Viewer

Options

266
Wireshark

267
Wireshark

• Freeware packet sniffer application enabling you to view traffic passed over the
network
• Advantages:
• Used for live/offline network troubleshooting and analysis
• Strong filtering
• SIP Call flow and Play sound
• And more
• AudioCodes add advance filtering for DTM/DSP debug

268
Capture Interfaces
• Capture > Options…
• Select the network interface currently used by the computer

269
Capture Output & Options

270
Wireshark Main Window

Filter Bar

Packet list
pane

Packet details pane

Packet bytes
pane

271
Coloring Rules
• Assign a color to each protocol to facilitate quick analysis
• Define general rules e.g., TCP, UDP at the bottom of the coloring list because processing is
from top to bottom until a match is found

272
Generating Call Flow
• Visually represents entire call flow
• Telephony > VoIP Calls

273
Playing G.711 RTP Stream

274
Analyzing RTP Data Stream
• Extracts audio from data packets (G.711 only)

275
SIP Calls Tests
Testing SIP calls
• The SIP test call simulates a complete SIP signaling process.
• Setup and registration of calls

• A simulated endpoint can be configured on the device to test SIP signaling of calls between it
and a remote destination
• Tests involve both incoming and outgoing calls

• Useful for remote verification of SIP message flow without involving the remote end side in a
debug process
• As any other call, a test call sends Syslog messages to a Syslog server, showing the SIP
message flow, DTMF signals, termination reasons, as well as voice quality statistics.

277
Testing SIP calls

• Test calls can be dialed automatically at a user-defined interval and/or manually

when required
• The simulated phone and remote endpoints are defined as SIP URIs (user@host)
• The remote destination can be defined as an IP address, or according to an
Outbound IP Routing rule
• Automatic registration of the endpoint can also be enabled

278
DTMF Tones configuration – (v6.6)
• By default, the DTMF signal that is played to answered incoming and outgoing test calls is
"3212333“
• This can be changed by using the GUI or ini file parameter, TestCallDtmfString
• Note: To generate DTMF tones, the device's DSP resources are required

279
Incoming test calls configuration – (v6.6)

280
DTMF Tones configuration – (v7.2)
• By default, the DTMF signal that is played to answered incoming and outgoing test calls is
"3212333“
• This can be changed by using the GUI or ini file parameter, TestCallDtmfString
• Note: To generate DTMF tones, the device's DSP resources are required

281
Incoming test calls configuration – (v7.2)

282
Incoming test calls configuration
• The Basic Test Call feature tests incoming Gateway calls from a remote SIP endpoint to a simulated test
endpoint on the device
• The only required configuration is to assign a prefix number (test call ID) to the simulated endpoint
• All incoming calls with this called (destination) prefix number is identified as a test call and sent to the
simulated endpoint
• The Basic Test Call feature tests incoming calls only and is initiated only upon receipt of incoming calls
with the configured prefix.

Remote SIP UA
Device Device
Network

Called Number: Prefix “450”

Simulated test point
Test Call ID: “450”

283
Outgoing test calls configuration – (v6.6)

284
Outgoing test calls configuration – (v6.6)

defined as a user or user@host

defined as a user or user@host

GW Tel2IP
IP Group
Dest Address

GW & IP2IP
SBC

Disable (default)
Enable

285
Outgoing test calls configuration – (v6.6)

Maximum number of concurrent channels for the test

session. For example, if endpoint "101“ is configured
Caller (default)
and the parameter is set to "3", the device
Called
automatically creates simulated endpoints - "101",
"102" and "103“
Number of Calls
Per Second 0 means infinite
-1 means that the value is calculated according to the
values of the 'Calls Per Second' and 'Maximum
Channels for Session' parameters

Once = The test runs until the lowest

value between the following is 0 means infinite
reached: The valid value is 0 to 100000
• Maximum channels
• Call duration Interval (in minutes) between
• Test duration expires automatic outgoing test Calls
Continuous = The test runs until the DTMF (default) The valid value range is 0 to 100000
configured test duration is reached PRT Default is 0 = scheduling is disabled

286
Test calls starting and stopping – (v6.6)
1. In the Test Call table, select the required test call entry; the Actions button appears above the table.
2. From the Actions drop-down list, choose the required command:
• Dial to start the test call
• Drop Call to stop the test call
• Restart ends all established calls and then starts the test call session again

2
3

287
Test calls viewing summary and statistics – (v6.6)
1. Select the test call table entry whose call statistics you want to view.
2. Click the Show/Hide button; the call statistics are displayed in the Test Statistics pane located beneath
the table

288
Outgoing test calls configuration – (v7.2)

289
Outgoing test calls configuration – (v7.2)

defined as a user or user@host

defined as a user or user@host

GW Tel2IP
Group
Dest Address

GW Tel2IP
SBC

Disable (default)
Enable

290
Outgoing test calls configuration – (v7.2)

Caller (default)
Called

Maximum number of concurrent channels for the test

session. For example, if endpoint "101“ is configured and
the parameter is set to "3", the device automatically creates
Number of Calls simulated endpoints - "101", "102" and "103“
Per Second

0 means infinite
-1 means that the value is calculated according to the values
of the 'Calls Per Second' and 'Maximum Channels for
Session' parameters

Once = The test runs until the

lowest value between the
following is reached:
• Maximum channels 0 means infinite
• Call duration The valid value is 0 to
• Test duration expires 100000
Continuous = The test runs
until the configured test Interval (in minutes) between automatic outgoing test Calls
duration is reached The valid value range is 0 to 100000
DTMF (default) Default is 0 = scheduling is disabled
PRT

291
Test calls starting and stopping – (v7.2)
1. In the Test Call table, select the required test call entry; the Actions button appears above the table.
2. From the Actions drop-down list, choose the required command:
• Dial to start the test call
• Drop Call to stop the test call
• Restart ends all established calls and then starts the test call session again

2
3

292
Test calls viewing summary and statistics – (v7.2)

293
Hands-on Lab 1

Basic Call
Hands-on Lab 2

SIP Call Tests

Hands-on Lab 3

Call with Proxy

Lesson 11

MP-11x FXO
FXO Gateways
• FXO (Foreign Exchange Office) – Generates the on-hook and off-hook indicators used to
signal a loop closure at the FXS's end of the circuit
• Analog telephone handsets, fax machines and (analog) modems are FXO devices
• FXO gateways convert (in real time) loop start signaling to SIP and variable electric current to
RTP

FXO Gateway

PBX

IP Signaling
IP Phone
IP Voice
IP
Local Loop

298
PBX to IP Calls (Tel2IP)

• The FXO gateway provides the following operating modes for Tel-to-IP calls:
• Automatic Dialing
• When a call is received from the PBX, the gateway automatically dials a preconfigured telephone
number to IP
• Hot Line
• When a call is received from the PBX, and no digit is dialed for HotLineToneDuration, the
gateway automatically dials a preconfigured telephone number to IP
• Collecting Digits
• When a call is received from the PBX, the gateway answers the call and plays a second dial tone
to the PBX, the user on the PBX side then dials the number they wish to reach

299
Automatic Dialing

• Defines telephone numbers that are automatically dialed when a specific port is
used
• Available options:
• Enable: the number in the 'Destination Phone Number' field is automatically dialed if a
ring signal is detected on a port
• Hotline: When ring is detected and no digit is dialed for HotLineToneDuration, the
number in the 'Destination Phone Number' field is automatically dialed
• Disable: Automatic dialing option on the specific port is disabled

300
Automatic Dialing (v6.6)

301
Automatic Dialing (v7.2)

302
Automatic Dialing

• Auto Dial Status = Enable

• The FXO receives call from the PBX
PBX FXO Gateway IP Phone
• If the CallerID recognizes the FXO, sends
SIP Invite immediately
FXO detects rings on line

• If the CallerID does not recognize the FXO, Invite sent immediately
the FXO sends the SIP Invite after 0-2
rings, based on the parameter 18x

‘RingsBeforeCallerID’
200 OK

The FXO off-hook

303
Automatic Dialing

• Auto Dial Status = Hotline

• The FXO receives call from the PBX
PBX FXO Gateway IP Phone
• After 2 rings the FXO off-hook s and sends
FXO detects rings on line
dial tone to the PBX
• The FXO recognizes CallerID based on the The FXO off-hook
after 2 rings

parameter ‘RingsBeforeCallerID’ Plays Dial Tone on line

• If after a period of time (based on the

parameter ‘HotLineToneDuration’) no
Waiting for digits
digits arrive, the FXO sends a SIP Invite to from PBX
the Auto Dial number
Invite
• The FXO sends SIP Invite to the IP Phone

304
Collecting Digits Dialing

• Auto Dial Status = Disable

• The FXO receives a call from the PBX
PBX FXO Gateway IP Phone
• After 2 rings the FXO off-hooks and sends
FXO detects rings on line
a dial tone to the PBX
• The FXO recognizes the CallerID based on The FXO off-hook
after 2 rings

the parameter ‘RingsBeforeCallerID’ Plays Dial Tone on line

• The FXO collects the digits that were sent

from the PBX
collects digits from
• The FXO sends a SIP Invite to the IP Phone PBX

Invite

305
PBX to IP Calls (Tel2IP) – Auto dial example

• The PBX ext’ 100 called to the Hunt Group number 9

• The FXO recognizes the CallerID (100) PBX 9

• The Auto number is 200 100

• The FXO sends SIP invite to the IP Phone Hunt Group

9
Off-Hook
• The IP Phone sends back 200 OK
FXO
• The FXO sends Off-Hook to the PBX 200
Gateway

LAN

OK

Auto Dial 200

HotLine

306
PBX to IP Calls (Tel2IP) – Collecting Digits example

• The PBX ext’ 100 calls to the Hunt Group number 9

• The FXO sends Off-Hook to the PBX PBX 200
9

•
100
The FXO sends Dial Tone to the PBX
Hunt Group
• The FXO recognizes the CallerID (100) 9
200 Off-Hook + Dial Tone
• The PBX ext’ dial 200 FXO
Gateway
• The FXO sends SIP invite to the IP Phone 200

• The IP Phone sends back 200 OK LAN

OK

200
Collecting
Digits

307
IP to PBX Calls (IP2Tel)

• The FXO gateway provides the following operating modes for IP-to-Tel calls:
• One stage dialing:
• When a new call is received from the IP, the FXO off-hooks the PBX line connected to the
telephone, and immediately dials the destination telephone number

• Two stage dialing:

• When the IP caller is required to dial twice
• The caller initially dials to the FXO gateway and only after receiving a dial tone from the PBX (via
the FXO gateway) dials the destination telephone number

308
IP to PBX Calls (IP2Tel) (v6.6)

309
IP to PBX Calls (IP2Tel) (v7.2)

310
IP to PBX Calls (IP2Tel) – One stage dialing
• The FXO receives an Invite from IP Phone
• The FXO sends a 100 Trying
• The FXO seizes the line from the PBX PBX FXO Gateway IP Phone

• The FXO waits for a PBX dial tone

Invite
• The FXO receives a dial tone from the PBX 100 Trying
FXO seizes line
• The FXO sends 183 to the IP Phone and opens the
Voice channel Wait for Dial Tone
183

• The FXO sends the DTMF to the PBX Voice Channel

• The FXO receives connect from the PBX Dial Tone

• The FXO sends 200 OK to IP Phone FXO Sent DTMF

Connect
200 OK

One stage

311
IP to PBX Calls (IP2Tel) – Two stage Dialing
• The FXO receives Invite from
IP Phone
• The FXO sends 100 Try
PBX FXO Gateway IP Phone
• The FXO seizes line from the PBX
• The FXO waits for PBX dial tone Invite

100 Try
• The FXO sends 200 OK to the IP Phone and opens FXO seizes line

the Voice channel 200 OK

• The IP Phone receives dial tone from the PBX Wait for Dial Tone
Voice Channel
• The IP Phone sends the DTMF to the PBX
Dial Tone

DTMF

Two stage

312
Call Termination on FXO

• The FXO can be configured to detect the following disconnect indications:

• Detection polarity reversal/current disconnect
• Detection of Reorder, Busy, Dial, or Special Information Tone (SIT) tones
• Note that this method requires the correct tones frequencies and cadence to be defined in the
Call Progress Tones file

• Special DTMF code

• A digit pattern when received from the Tel side, indicates to the gateway to disconnect the call

313
Call Termination on FXO

• The PBX doesn't disconnect the call; however instead signals to the gateway that
the call is disconnected

Active Call between IP user to PBX user trough FXO

PBX user disconnect the call

PBX should sign the FXO

Supervision Signals

BYE

PBX User PBX FXO IP User

314
FXO Settings (v6.6)

315
FXO Settings (v7.2)

316
Hands-on Lab 4

FXO
Lesson 12

Common Features
Routing
Routing Tables (reminder)

• 2 routing tables for incoming and outgoing calls:

• Outbound IP Routing Table
• Up to 50 for MPs and up to 180 for Mediants Tel-to-IP/outbound IP call routing rules
• The gateway uses these rules to route calls from Tel to IP

• Inbound IP Routing Table

• Up to 24 for MPs and up to 120 for Mediants inbound call routing rules
• Used for:
• IP to Tel routing / IP to Hunt Group
• IP to IP routing

• Routing can be performed before or after manipulation rules are applied

320
IP to Tel (Hunt) Routing Table

The IP to Tel Routing Mode parameter

determines the order between routing
calls to Hunt/Trunk Groups and
manipulation of the number

Match Area Action Area

• The table is divided to main areas: Match characteristics and Action to take
• If the incoming call matches the characteristics of a rule, then the call is sent to the destination configured for that rule
• ‘-1’ in the table refers to “Not Configured”
321
Tel to IP Routing Table

The Tel to IP Routing Mode

parameter determines the order
between routing calls to IP and
manipulation of the number

Match Area Action Area

• The table is divided to main areas: Match characteristics and Action to take
• If the incoming call matches the characteristics of a rule, then the call is sent to the destination configured for that rule
• ‘-1’ in the table refers to “Not Configured”
322
Numbers Notation for Routing and Manipulation

Flexible numbers notations for describing the prefix and/or suffix source and/or destination phone
numbers and SIP URI user names:
• Prefix [n-m] or Suffix (n-m)
• Represents a range of numbers Destination Phone Prefix Source Phone Prefix

• Prefix [n,m,...] or Suffix (n,m,...) 1 9x*

• Represents multiple numbers 2[2,6,7,9] 1xxx

• Multiple ranges such as [n-m,s-t] are also supported 2[1-4,7,9] 1xxx#

• Up to three digits can be used to denote each number [100-150,222,244,300-499] 1*

• x (letter ‘x’) 6[100-300] (99)

976(99) 2[1-4]
• Represents any single digit
6[100-300]# *
• # (Pound symbol)
* *
• Represents the end of a number
• * (asterisk symbol)
• Represents any number
323
Numbers Notation
• Examples:
• [5200-5300]#
• represents all numbers from 5200 to 5300
• [2,3,4]xxx#
• represents four-digit numbers that start with 2, 3 or 4 (2000-4999)
• 54324
• represents any number that starts with 54324
• 54324xx#
• represents seven-digit numbers that start with 54324
• 123[100-200]#
• represents six-digit numbers that start with 123 (123100 to 123200)
• (100)
• represents any number that finishes with 100
• (266[1-9])
• represents any number that finishes with 2661 to 2669
324
Alternative Routing
Alternative Routing

• Enables reliable routing of Tel to IP calls

• Enables the system to constantly check the availability of connectivity and suitable
Quality of Service (QoS) before routing
• If the alternative routing destination is the gateway itself, the call can be configured
to be routed back to one of the gateway's trunk groups and thus, back into the
PSTN (PSTN Fallback for digital gateways)
• Can also be applied when Proxy is used (RedundantRoutingMode = 1)

326
Triggering the Alternative Routing

• Loss of connectivity (monitored by ICMP ‘Ping’)

• Inappropriate level of QoS is detected – delay or packet loss calculated according to
previous calls (RTCP statistics)
• DNS host name cannot be resolved
• Relevant parameters:
• AltRoutingTel2IPEnable
• AltRoutingTel2IPMode
• A pre-defined ‘Reasons for Alternative Routing’ rule is matched

327
Adding an Alternative Route

• In the Tel to IP Routing table,

add an entry with the same
prefix to a different IP Address
• Or enter a FQDN that maps to
two separate IP Addresses
• Applies to either software
release (v6.6/v7.2)

328
Reason for Alternative Routing (v.6.6)

Two tables are used to

force routing to an The release reason for IP
alternate address: to Tel calls is provided in
• IP to Tel Q.931 notation
• Tel to IP
The release reason for
Tel to IP calls is
provided in SIP 4xx, 5xx,
and 6xx response codes
Each group enables
you to define up to
5 different release
reasons

329
Reason for Alternative Routing (v7.2)

• Two tables are used to force routing to an alternate address:

• IP to Tel (Hunt/Trunk)
• Tel to IP
• Each group enables you to define up to 20 (v7.2) or 5 (pre v6.8) different release
reasons
• The release reason for Tel to IP calls is provided in SIP 4xx, 5xx, and 6xx response
codes
• The release reason for IP to Tel calls is provided in Q.931 notation

330
Reason for Alternative Routing IP-to-Tel (v7.2)

331
Reason for Alternative Routing Tel-to-IP (v7.2)

Generic category of
statuses can be
defined

A specific status
code can be defined

332
Number Manipulation
Number Manipulation

• Number Manipulation tables for incoming and outgoing calls are provided
• Used to modify Destination and Source telephone numbers so that calls can be
routed correctly
• Manipulation can occur before or after a routing decision is made
• Using Manipulation Tables you can:
• Allow/Restrict Caller ID information (Source Number for Tel-to-IP Calls)
• Assign NPI/TON to IP-to-Tel calls
• Optionally run a second (additional) ‘round’ of number manipulations for
IP-to-Tel calls on an already manipulated number

334
Number Manipulation v6.6

335
Number Manipulation v6.6

336
Number Manipulation: Rules and Actions (v6.6)
• Destination Phone Number Manipulations for IP to Tel calls

Matching Rules

1. Destination (called) telephone number prefix and/or suffix

2. Source (calling) telephone number prefix and/or suffix
3. Represents the source IP address of the caller (obtained from
the Contact header in the INVITE message)
4. Defines the URI host name prefix of the incoming SIP INVITE
message in the From header
5. Defines the Request-URI host name prefix of the incoming SIP
INVITE message

Actions to Take
1. Action 1: Number of digits to remove from left of the number
2. Action 2: Number of digits to remove from right of the number
3. Action 3: Number of digits to leave from the right
4. Action 4: Number or string to add to the left of the number
5. Action 5: Number or string to add to the right of the
6. Enables caller ID

337
Number Manipulation: Rules and Actions (v6.6)
• Source Phone Number Manipulations for IP to Tel calls

Matching Rules

1. Source (calling) telephone number prefix and/or suffix

2. Represents the source IP address of the caller (obtained from
the Contact header in the INVITE message)
3. Defines the URI host name prefix of the incoming SIP INVITE
message in the From header
4. Destination (called) telephone number prefix and/or suffix
5. Defines the Request-URI host name prefix of the incoming SIP
INVITE message

Actions to Take
1. Action 1: Number of digits to remove from left of the number
2. Action 2: Number of digits to remove from right of the number
3. Action 3: Number of digits to leave from the right
4. Action 4: Number or string to add to the left of the number
5. Action 5: Number or string to add to the right of the
6. Enables caller ID

338
Number Manipulation: Rules and Actions (v6.6)
• Destination Phone Number Manipulations for Tel to IP calls

Matching Rules
1. Destination telephone number prefix
2. Represents the source IP address of the call (obtained from
the Contact header in the INVITE message)
3. Defines the source Hunt Group ID. To denote all Hunt Groups,
leave this field empty
4. Defines the IP Group from where the IP call originated

Actions to Take
1. Action 1: Number of digits to remove from left of the number
2. Action 2: Number of digits to remove from right of the number
3. Action 3: Number of digits to leave from the right
4. Action 4: Number or string to add to the left of the number
5. Action 5: Number or string to add to the right of the
6. Defines the NPI (Numbering Plan Indicator)
7. Defines the TON (Type of Number)
8. Enables caller ID
339
Number Manipulation: Rules and Actions (v6.6)
• Source Phone Number Manipulations for Tel to IP calls

Matching Rules
1. Destination telephone number prefix
2. Represents the source IP address of the call (obtained from
the Contact header in the INVITE message)
3. Defines the source Hunt Group ID. To denote all Hunt Groups,
leave this field empty

Actions to Take
1. Action 1: Number of digits to remove from left of the number
2. Action 2: Number of digits to remove from right of the number
3. Action 3: Number of digits to leave from the right
4. Action 4: Number or string to add to the left of the number
5. Action 5: Number or string to add to the right of the
6. Defines the NPI (Numbering Plan Indicator)
7. Defines the TON (Type of Number)
8. Enables caller ID

340
Number Manipulation: Rules and Actions (v6.6)

• Gateway first attempts to match patterns at the top of the table More specific rules
should be at the top and more generic ones at the bottom
• Applies to either software release (v6.6/v7.2)

‘551’ will never match

because ’55’ matches every
prefix that starts with ’55’

341
Numbers Manipulation (examples)

If the destination number is 97234100, it’s transformed into 8884100

If the destination number is 97246000, it’s transformed into 9994600

342
Numbers Manipulation (examples)

If the destination number is 19764, it’s transformed into 9991976

If the destination number is 3541, it’s transformed into 88835

343
Numbers Manipulation (examples)

If the destination number is 9764506000, it’s transformed into 060

If the destination number is 97239763076000, it’s transformed into 4760
If the destination number is +97239763076000, it’s transformed into 40760
If the destination number is 9764076, it’s transformed into 9764076 344
Profiles
Profiles

• A Profile is a set of configuration parameters

• Profiles provide high-level adaptation when connected to a variety of equipment
(from both Tel and IP sides), each of which requires different system behavior
• Using Profiles, users can assign different Profiles (behavior) on a per-call basis, or
associate different Profiles to the gateway’s endpoints/B-channels
• Profiles contain parameters, such as Coders, T.38 Relay, Voice and DTMF Gains and
more
• Each call can be associated with one or two Profiles, Tel Profile and/or IP Profile
• If both IP and Tel profiles apply to the same call, the preferred Profile is determined by
the Preference option
• If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are
applied

346
Facilitating Support for Coders

• Software Upgrade Key only includes coders G.711 and G.726 (by default)
• To facilitate gateway support for other coders (e.g., G.723.1):
• Update the gateway’s Software Upgrade Key
• MP-1xx have no Feature Key
• Coder Group Table
• Allows you to configure up to 10 coders for the Gateway
• The first coder in the list has the highest priority
• A coder can appear only once in the table
• The Packetization Time determines how many coder payloads are combined into a single RTP
packet
• The Gateway always uses the packetization time requested by the remote side for sending RTP packets
• Enable/Disable the Silence Suppression option per coder
347
Coder Group Table (v6.6)

348
Coder Groups Settings (v6.6)
• Use the ‘Coder Group Settings’ screen to defines up to 4 different coder groups
• These Coder Groups are used in the Tel Profile and/or IP Profile Settings screens to assign
different coders to Profiles
• Each Coder Group contains up to 10 coders

349
Coder Group Table (v7.2)

350
Tel Profile (v6.6)
• Can define up to 9 different Tel Profiles
• These Profiles are used in the 'Endpoint Phone Number Table' screen where they can be
assigned to the gateway's channels

351
Tel Profile (v7.2)

352
Assigning Tel Profile to Endpoint (v6.6)

• Tel Profile can be assigned to Endpoints/B-Channels

353
Assigning Tel Profile to Endpoint (v7.2)

354
IP Profile (v6.6)
• Can define up to 9 different IP Profiles
• These Profiles are used in the 'Tel to IP Routing' and 'IP to Trunk Group Routing Table' screens for
associating IP Profiles to routing rules. IP Profiles can also be used when working with a Proxy Server

355
IP Profile (v7.2)

356
Assigning IP Profile to Routing Rules (v6.6)

• Tel to IP Routing
• Affects the calls that are sent from the Gateway

357
Assigning IP Profile to Routing Rules v7.2

358
Assigning IP Profile to Routing Rules (v6.6)

• IP To Hunt Group Routing

• Affects the calls that are received by the Gateway
• SIP messages that arrive without matching SIP parameters, are dropped

359
Assigning IP Profile to Routing Rules (v7.2)

360
DTMF Transport
DTMF

• Dual Tone Multi-Frequency

• Pressing a key on the phone's keypad generates two simultaneous tones, one for
the row and one for the column
• These are decoded by the exchange to determine which key was pressed

362
DTMF Transport Types

• In-Band
• Transparent – DTMF digits are carried within the voice stream using a high bit rate coder
• RFC 2833
• Out-of-Band using SIP signaling (in this mode DTMF digits are erased from the RTP
stream)
• INFO (Nortel) – Each INFO message can carry more than 1 digit (buffering digits)
• INFO (Cisco) – Each INFO message carries 1 digit
• INFO (Korea)
• NOTIFY

363
RFC 2833

• In this mode, DTMF digits are carried to the remote side as part of the RTP stream
in accordance with RFC 2833 standard
• To enable this mode, define the following:
• TxDTMFOption = 4
• RxDTMFOption = 3 (Declare RFC 2833 in SDP = Yes)
• To set the RFC 2833 payload type with a different value (other than its default 96) use the
parameter RFC2833PayloadType
• The Gateway negotiates the RFC 2833 payload type using local and remote SDP and
sends packets using the payload type from the received SDP

364
DTMF using INFO messages

• In this mode, DTMF digits are carried to the remote side within INFO messages
• TxDTMFOption should be set according to the selected method
• Cisco
• Nortel
• Korea

• RxDTMFOption = 0

365
DTMF settings (v6.6)
• To enable this mode, define the following:
• TxDTMFOption = 4
• RxDTMFOption = 3 (Declare RFC 2833 in SDP = Yes)
• To set the RFC 2833 payload type with a different value (other than its default 96) use the parameter
RFC2833PayloadType

366
DTMF settings (v7.2)

367
Digit Collection Rules
Digit Collection Rules

• When dialing ends, the gateway uses the collected digits for the called destination
number. Dialing ends when:
• The maximum number of digits is dialed (configured by the parameter MaxDigits)
• The Inter-digit Timeout expires (configured by the parameter TimeBetweenDigits)
• The '#' key is dialed (to allow '*' and '#' to be used for telephone numbers set
IsSpecialDigits=1)
• A digit map pattern is matched (using the parameter DigitMapping or the Dial Plan file)
• Using the dial plan file, it is possible to assign different dial plans to different ports (using Tel
profiles)

369
DTMF settings v6.6
• Determining when dialing is complete
• The maximum number of digits is dialed
• The Inter-digit Timeout expires
• Special Digit Mapping
• A digit map pattern is matched/Dial Plan is used

370
DTMF settings (v7.2)

371
Hands-on Lab 5

Hunt Group
Hands-on Lab 6

Number Manipulations
Lesson 13

Mediant Family
Digital Gateways Overview

• Digital PRI and BRI VoIP gateways

• SBC capability (some of them)
• Up to 16,000 simultaneous calls
• Gateway types:
• Small: Mediant 500L, Mediant 500, Mediant 800
• Medium: Mediant 1000B
• Large: Mediant 3000, Mediant 5000, Mediant 8000*
Note:
• The latest maintenance firmware version for Mediant 5000 and 8000 is 6.6
• The latest maintenance firmware version for Mediant 3000 is 7.0

Mediant 500L Mediant 500 Mediant 800 Mediant 1000B Mediant 3000
375
Mediant 500L / 500 / 800B /800C/ 1000B* MSBR

• Dual Processors (CMX & RMX)

• WAN port – WAN Gigabit Ethernet, T1 WAN, SHDSL, ADSL/VDSL
• Strong CLI management
• Data Routing capabilities by providing static routing and dynamic routing protocols
such as RIP/OSPF and BGP
• Supports a selection of WAN interfaces providing flexibility connecting to Service
Providers
• Firewall
• QoS
• Mediant 500L/500 and 800 only: 3G connection (using USB 3G stick) used as
primary WAN interface or as optional/backup when primary WAN fails
Note: The latest maintenance firmware version for Mediant 1000B MSBR is 7.0
376
Mediant 500L Gateway and E-SBC

• Networking device combining multiple service functions

• Enterprise Class Session Border Controller (E-SBC)
• LAN Ethernet ports:
• Four Gigabit Ethernet LAN ports
• Optional PSTN interfaces:
• Up to 4 BRI
• Up to 4 FXS/FXO
• OAM&P:
• Embedded HTTP/S-based Web Server
• Command Line Interface (CLI)
USB port
• Configuration ini file
• SNMP RS-232 serial Four Gigabit
communication Ethernet ports
• REST API
377
Mediant 500L GW/E-SBC/MSBR
• Optional telephony connections:
• 4 BRI
• 4 FXS
• 4 FXO
• Data capability:
• Four Gigabit Ethernet (10/100/1000Base-T) LAN ports
• WAN port – Single copper GE /SFP, ADSL2+ / VDSL2
• 3G connection (using USB 3G stick) used as primary WAN interface or as optional/backup when
primary WAN fails
• Firewall (MSBR only)
• Route (MSBR only)
• QoS (MSBR only)

378
Mediant 500 Gateway and E-SBC

• Networking device combining multiple service functions

• Enterprise Class Session Border Controller (E-SBC)
• LAN Ethernet ports :
• Four Gigabit Ethernet (10/100/1000Base-T) LAN ports
• PSTN connectivity
• Up to 1 E1/T1/J1 trunk
• OAM&P:
• Embedded HTTP/S-based Web Server
• Command Line Interface (CLI)
• Configuration ini file
• SNMP Two USB ports RS-232 serial four Gigabit
communication Ethernet ports
• REST API
379
Mediant 500 Gateway/E-SBC/MSBR

• Optional telephony connections:

• 1 PRI
• 2 BRI
• 4 FXS
• 1 FXO

• Data capability:
• Four Gigabit Ethernet (10/100/1000Base-T) LAN ports
• WAN port – Single copper GE /SFP, ADSL2+ / VDSL2
• 3G connection (using USB 3G stick) used as primary WAN interface or as optional/backup when
primary WAN fails
• Firewall (MSBR only)
• Route (MSBR only)
• QoS (MSBR only)
380
Mediant 500 Gateway/E-SBC/MSBR (Front View)

WiFi 802.11a/b/g/n
2x2 MIMO Dual band 2.4GHz/5GHz

RJ-45 port for RS-232

serial communication

WiFi Enable WAN interfaces LAN interfaces Telephony interfaces

button Options: ▪ 4 x Gigabit Ethernet Options (4 voice channels):
2 x USB2.0 ▪ GE Copper ▪ 4 FXS
▪ 3G/4G mobile WAN modem ▪ 100/1000 SFP ▪ 3 FXS + 1 FXO
▪ External USB hard drive or ▪ ADSL2+/VDSL2 ▪ 2 BRI
flash disk (storage) ▪ SHDSL*
▪ T1/E1* Dimensions:
4.37 (1U) x 31.0 x 21.0 cm (1.72 x 12.2 x 8.3 in.)

Note: The figure above is used only as an example. The number and type of port interfaces depends on the ordered model.

381
Mediant 500 Gateway/E-SBC/MSBR (Front View)

1. Wi-Fi button for enabling and disabling Wi-Fi (available only if ordered with Wi-Fi)
2. Reset pinhole button for resetting the device and optionally, for restoring the device to
factory defaults
3. Console - RJ-45 port for RS-232 serial communication
4. WAN interface, can be any of the following: Copper GE, SFP module, ADSL/2+ and VDSL2
5. Up to 4 GE ports for connecting to LAN network
6. Telephony interfaces, depending on ordered configuration: 1 PRI E1/T1, up to 2 BRI ports, up
to 4 FXS ports, 1 FXO
7. LEDs indicating the status of the power, reboot/initialization, and Wireless LAN interface
8. 2 USB 2.0 ports, which can be used for the following: 3G cellular WAN modem for primary or
backup WAN and external USB hard drive or flash disk for USB storage capabilities

382
Mediant 500 Gateway/E-SBC/MSBR (Rear View)

1. Protective earthing screw

2. Power switch (O is off; I is on).
3. AC power supply entry
4. Wi-Fi antennas, providing wireless LAN 802.11 b/g/n access point at 2.4 GHz and 5 GHz, 2 Tx
and 2 Rx, enabling data rates of up to 300 Mbps

383
Mediant 800 Gateway and E-SBC
• Networking device combining multiple service functions
• Enterprise Class Session Border Controller (E-SBC)
• LAN Ethernet ports: Power / Status LEDs Reset pinhole
• Up to 4 Gigabit Ethernet button
• Up to 8 Fast Ethernet
FXS/FXO/BRI/E1/T1
• Integrated PSTN connectivity
• Up to 2 E1/T1/J1 trunks
• 8 BRI ports (16 calls)
• Up to 12 analog FXS/FXO ports
• OAM&P:
• Embedded HTTP/S-based Web Server
• Command Line Interface (CLI) Four Gigabit Eight Fast Ethernet
• Configuration ini file Ethernet LAN ports LAN ports
• SNMP
• REST API
• Integrated Open Solutions Network (OSN) server platform
384
Mediant 800 Gateway/E-SBC/MSBR

• The Mediant 800B MSBR is based on Mediant 800 MSBR,

but with additional support:
• 2 PRI
• 8 BRI
• 12 FXS/FXO

• Data capability:
• Up to 12 LAN ports Power over Ethernet (PoE)
• WAN port – Integral copper GE, plus optional 2 WAN interfaces (xDSL or GE UTP/SFP). T1/E1, SHDSL,
ADSL2+, VDSL, 100Base-X, 1000Base-X (SFP Format)
• Wi-Fi Access Point support for 802.11 a/b/g/n, dual band 2.4 GHz, 5GHz
• 3G connection (using USB 3G stick) used as primary WAN interface or as optional/backup when
primary WAN fails
• 2 x USB interface ports
• RJ-45 serial connector
385
New Mediant 800C

• Up to 4 x E1/T1

• Dual flash memory, allowing the user to revert to the previous software version
after a software upgrade failure

• Dual power supply – in addition the AC power supply supplied by default,

the chassis can be ordered with an option DC power supply inlet

• Increased Gateway and SBC session capacity due to powerful CPU

386
Standalone OSN Server Hosted on Mediant 800

Parameter OSN6 OSN7

Intel® Core™ i7-5850EQ Processor Intel® Pentium® Processor D Series
CPU 4 Cores, 6M Cache, 2.7 GHz 2 Cores, 3M Cache, 2.60 GHz
Memory 32 GB 16 GB

Hard Drives 128 GB SSD (or higher, for special request)

• 2 Gigabit Ethernet external (rear panel)

• 1 Gigabit Ethernet internal bus, connected to the Mediant
Interfaces
• 3 USB 2.0
• VGA

387
Mediant 1000B Gateway and E-SBC
• LAN Ethernet ports
• Up to 3 Pairs of 1+1 LAN interfaces
• Modular – can host a variety of interfaces MSBR: CRMX
• 1 to 6 E1/8T1/ trunks (up to 192 channels) SBC: CMX
• 4 to 20 BRI ports (40 calls) Field-Replaceable
• 4 to 24 analog (FXS/FXO) ports FXO/FXS/Trunks/BRI/MPM Fan Tray Module
• Up to 4 MPMs for media processing Modules
• Enterprise Class Session Border Controller (E-SBC)
• Single or Dual Power Supply
• 2 OSN servers (Optional)
• OAM&P:
• Embedded HTTP/S-based Web Server
• Command Line Interface (CLI) 2 Power Supply
Serial Port Modules
• Configuration ini file
• SNMP
• REST API

388
LAN switching extension module

• CRMX module provides 3 LAN 10/100/1000 Ports

• By default, each couple configured in 1+1 redundancy mode
• Can be configured as individual ports (no redundancy)
• The SWX LAN Expansion module provides additional 4 LAN 10/100/1000 Ports
• Each group contains two ports, 1+1 redundancy
• Can be configured as individual ports (no redundancy)

389
Analog FXS / FXO Modules

• Modularity: 1-6 modules of 2 or 4 FXS/FXO ports (Up to 24 ports)

• Hot swappable

390
Digital PRI Module

• E1/T1/J1 capabilities
• 1, 2, or 4 port configurations.
• Since release 6.8 up to 6 E1 ports or 8 T1 ports
• Previous releases supported up to 4 E1/T1 ports
• PSTN Fallback support
• Hot swappable

391
BRI Module

• UP to 4 ports of S/T Interface

• Point-to-Point or Point-to-Multi-Point configuration
• Each port can be NT or TE (Network side or User side)
• Up to 8 VoIP sessions per module
• Up to 5 modules per Mediant 1000 (up to 20 BRI ports)

392
MPM Module

• Used for IPmedia functionalities (e.g., conferencing and for playing

announcements)
• Can also be used for additional DSP resources (e.g., for IP to IP)
• Up to 4 MPM modules are supported in a single chassis
• Releases prior to version 6.8 supported up to 3 MPM modules
• An MPM module provides 40 IPmedia channels
• For conferencing capabilities, an MPM module must be housed in Slot #6. In this
case, the number of DSP channels in this module is reduced to 20

393
Standalone OSN Server Hosted on Mediant 1000B

Parameter OSN3C OSN4B

Intel® Pentium® Processor D1508 Intel® Xeon® Processor D-1527
CPU 2 Cores, 3M Cache, 2.20 GHz 4 Cores , 6M Cache, 2.20 GHz
RAM Memory 8 GB 16 GB

Hard Drives Up to 2 hard drives (HDMX modules) 500 GB HDD or 120GB SSD (2 HDD can work in Raid1)

• 2 Gigabit Ethernet external (rear panel)

• 1 Gigabit Ethernet internal bus, connected to the Mediant
Interfaces • USB 2.0
• RS-232
• Graphics
394
Digital Lifeline
• PSTN Fallback (Digital):
• If power fails or there is a loss of IP network connectivity, a relay connects trunks 1 to 2
and/or 3 to 4 in the same module
• To provide the link, a metallic switch inside the module closes so that the trunk from the
PBX is routed from the module to the PSTN

Trunk

FXS

• Lifeline (Analog):
• Lifeline is provided only by Port 1 on an FXS module
395
Lesson 14

Mediant Family Configuration

Lesson Objectives

• After completing this lesson you will:

• Know how to configure the PSTN interface

• Understand the PSTN ↔ Routing Table relationship

397
Configuring the TDM Bus

• TDM Bus Clock Source (Network/Internal)

• Clock source on which the gateway synchronizes

• TDM Bus PSTN Auto FallBack Clock

• Relevant only if TDMBusClockSource = Network)
• Disable = Recovers the clock from the E1/T1 line defined by parameter ‘TDM Bus Local Reference’
• Enable = Recovers the clock from any connected synchronized slave E1/T1 line

• TDM Bus Local Reference

• Determines the Trunk ID used to synchronize the gateway’s clock when using external clock

• PCM Law Select (A-law/µ-law)

• Usually A-Law for E1 and µ-Law for T1

398
Configuring the TDM Bus

399
Configuring Key Trunk Parameters

• Protocol Type
• Sets the PSTN protocol to be used for this trunk
• If ‘Protocol Type’ of all PRI trunks displays 'None', select the protocol type (E1/T1) for a single
trunk and reset the gateway
• Only after the reset you will be able to continue configuring the trunks
• Clock Master
• Determines Tx clock source of E1/T1 line
• Recovered (0) = Generate clock according to Rx of E1/T1 line
• Generated (1) = Generate clock according to internal TDM bus
• ISDN Termination Side
• User side = ISDN User Termination Side (TE)
• Network side = ISDN Network Termination Side (NT)
• Select 'User side' when the PSTN or PBX side is configured as 'Network side’ and
vice-versa
400
Configuring Key Trunk Parameters

401
Digital Trunk Points of Information
• All Trunk spans must be of the same Line Type (all E1 or all T1)

• Different flavors of same Line Type (E1/T1) can be configured on available Trunks
(e.g., E1 Euro ISDN and E1 QSIG)

• Trunks are referenced in ini file and Syslog messages from ‘0’ regardless of whether
physical Trunks are numbered from ‘1’

E1
E1
Euro
QSIG
ISDN

402
Examples of Basic Trunk Issues

Why do I receive this message when I try to stop a trunk?

• The trunk can’t be stopped because it provides the gateway’s clock (assuming the
gateway is synchronized with the E1/T1 clock)

• Solution:
• Assign a different E1/T1 trunk to provide the gateway’s clock or enable ‘TDM Bus PSTN Auto
Clock’ in the 'TDM Bus Settings' screen

403
Examples of Basic Trunk Issues

Why do I have poor voice quality on all calls?

• Probably because the value you configured for the PCM Law Select parameter for the
Mediant is incorrect

• It must be identical to the value configured for the PCM Law Select parameter for the
PBX/PSTN

• A-law is usually used for E1 spans and µ-law for T1 spans

404
Trunk Group Table – E1/T1 and/or FXS

• Used to assign Trunk Groups, Profiles and logical telephone numbers to the
gateway's channels
• Trunks or B-Channels that are not defined are disabled

405
Trunk Group Settings
• Determines the method by which new calls are assigned to channels within each Trunk Group ID
• If such a rule doesn't exist (for a specific Trunk Group), the global rule defined by the Gateway General
Settings Channel Select Mode parameter applies

406
To change the protocol type between E1/T1 trunks

• To change the protocol type:

1. Save the current configuration (ini file) on your PC
2. Open the ini file in a txt editor (Notepad or similar)
3. Change the value of the ‘ProtocolType’ parameter to the required protocol type
4. Change the value of the ‘FramingMethod’ parameter according to the protocol type
you selected
5. Reload the modified ini file to the gateway

407
Configuring a Clock Option

• To use an internal gateway clock source:

• TDM Bus Clock Source = Internal
• Clock Master = Generated (for all gateway trunks)
• To recover clock source from PSTN/PBX:
• TDM Bus Clock Source = Network
• Clock Master = Recovered (for all ‘slave’ gateway trunks connected to PBX)
• To define the ‘slave’ trunk from which the gateway recovers its clock:
• TDM Bus PSTN Auto Clock = Enable
• The gateway automatically selects one of the connected ‘slave’ trunks
• TDM Bus Local Reference = Trunk #
• Configures a specific trunk

408
Verifying Trunk Synchronization

• To verify that the clock is synchronized and that there are no slips on the trunk,
access the CmdShell and enter these commands:
• pstn
• physical
• PstnGetPerformanceMonitoring trunknumber #
• Note that trunks are numbered 0 - 7

pstn

CAS/ PHysical/ PstnCOmmon/

/PStn>
ph

IsdnGetDChannelStatus PstnQueryTrunkStatus PstnSendAlarm PstnLoopCommands

PstnGetPerformanceMonitoring PstnStoPPerformanceMonitoring PstnStarTPerformanceMonitoring
/PStn/PHysical>
PstnGetPerformanceMonitoring 0 0

409
Verifying Trunk Synchronization (cont.)
/PStn/PHysical>
PstnGetPerformanceMonitoring 0 0

TrunkId = 0
Interval = 0
AlarmIndicationSignal = 0
LossOfSignal = 0
LossOfFrame = 0
FramingErrorReceived = 0
RemoteAlarmReceived = 0
LostCRC4multiframeSync = 0
CRCErrorReceived = 0
EBitErrorDetected = 0
BitError = 0
LineCodeViolation = 0
ControlledSlip = 0
ErroredSeconds = 0
ControlledSlipSeconds = 0
SeverelyErroredFramingSeconds = 0
SeverelyErroredSeconds = 0
BurstyErroredSeconds = 0
UnAvailableSeconds = 0
PathCodingViolation = 0
LineErroredSeconds = 0
DegradedMinutes = 0
AssessedSeconds = 427

410
Verifying Trunk Synchronization (cont.)

411
Loading a CAS File to the Gateway

• If CAS protocol is used, download a CAS state machine file to the gateway

412
Selecting CAS Table

413
Overlap Dialing
• Overlap dialing is a dialing scheme to send/receive called # digits in parts or several at once
• Opposite to en-bloc dialing in which the complete number is sent at once
• The gateway supports ISDN overlap dialing for incoming ISDN calls per E1/T1 trunk

None = Disable
Local receiving = the complete number is
sent in the INVITE Request-URI user part
Through SIP = each digit is sent to the IP
(based on RFC 3578)

414
Overlap Dialing

• The gateway stops collecting digits (for ISDN-to-IP calls) when:

• The sending device transmits a ‘sending complete’ IE in the ISDN Setup or the
subsequent INFO messages, to signal that no more digits will be sent
• The inter-digit timeout configured by the ‘TimeBetweenDigits’ parameter expires
(default = 4 seconds)
• The maximum allowed number of digits configured by the ‘MaxDigits’ parameter is
reached (default = 30 digits)
• A match is found with the defined digit map (configured by the ‘DigitMapping’
parameter)

415
Software Upgrade Key

•Supplied only with the digital gateways

•Determines features, capabilities and available resources
•Provided in string format or in a txt file to be loaded to the gateway
•Stored in the gateway's non-volatile flash memory
•Load key using either
• Web interface
• AudioCodes’ OVOC
• After loading the new key, the gateway must be physically reset

416
Software Upgrade Key

417
Routing Tables (Quick Review)

• 2 routing tables for incoming and outgoing calls:

• Outbound IP Routing Table
• Up to 180 Tel-to-IP/outbound IP call routing rules
• The gateway uses these rules to route calls from Tel to IP
• Inbound IP Routing Table
• Up to 120 IP-to-Tel/inbound call routing rules
• The gateway uses these rules to route calls from IP to Tel

• Routing can be performed before or after manipulation rules are applied

418
Routing Tables Syntax

• The table is divided 2 main areas: Match

characteristics and Action to take
• If the incoming call matches the
characteristics of a rule, then the call is sent
to the destination configured for that rule
• ‘-1’ in the table refers to “Not Configured”

419
Fields to Match

• Gateway attempts to match patterns at the top of the table first (first match)
• More specific rules should be at the top and more generic ones at the bottom

‘551’ will never match

because ’55’ matches every
prefix that starts with ’55’

420
Inbound IP Routing Table (IP2Tel)
• Used to route incoming IP calls to trunk groups

421
Inbound IP Routing Table (IP2Tel)

422
Outbound IP Routing Table (Tel2IP)
• Used to route outgoing calls from Tel to IP

423
Outbound IP Routing Table (Tel2IP)

424
Number Manipulation

425
Number Manipulation

Matching: Actions to Take:

1. Source IP Address 1. Stripped digits from left
2. Source Prefix 2. Stripped digits from right
3. Source Host Prefix 3. Number of digits to leave (from right)
4. Destination Prefix 4. Add Prefix
5. Destination Host Prefix 5. Add Suffix
6. Source IP Group 6. Add TON
7. Add NPI
426
Number Manipulation

427
Routing Mode Parameters

• The Tel to IP Routing Mode and IP to Tel Routing Mode parameters determine the order
between routing calls to Trunk Groups and manipulation of the number
• Route calls before manipulation (default)
• Route calls after manipulation

428
Hands-on Lab 7

Digital Gateways Basic

Hands-on Lab 8

Alternative Routing
Hands-on Lab 9

Profiles
Hands-on Lab 10

Primary Rate Interface

Lesson 15

Advanced Debugging Tools

Debug Recording

434
What is Debug Recording (DR)?

• A feature used to capture and record traffic sent and/or received by the device
• It is used for advanced debugging when you need to analyze internal messages and
signals
• The device can send debug recording packets to a debug capturing server
• Can record different types of traffic such as
• Media streams (RTP, T.38 and PCM)
• PSTN signaling (ISDN, CAS, SS7)
• Control messages (SIP, MGCP, MEGACO)
• Networking streams (such as HTTP and SCTP)
• Other internal information (such as DSP Events)

435
Debug Recording Advantages

• Can record all IP traffic sent by/received from the device

• Can record actual voice signal arriving from the TDM (before it enters the DSP)
• Useful for recording network traffic in environments where hub or port mirroring is
unavailable
• Useful for recording internal traffic between two endpoints on the same device
• Can include Syslog messages
• Debug Recording packets are captured using WireShark or a similar tool (requires
AudioCodes proprietary Plug-in)

436
Installing AudioCodes’ Proprietary Plug-in
1. Install Wireshark on your computer
• The Wireshark program can be downloaded from www.wireshark.org
2. Install the AudioCodes proprietary needed plug-in files as follows:
• Either download them from www.audiocodes.com/library/firmware?page=2
or copy them from your Student Kit USB Stick folder: \Utilities\Wireshark\Plugins_x64\*.*
• Then copy them to the directory in which you installed Wireshark, as follows:
\Wireshark\plugins\<Wireshark ver.>\epan\
3. Start Wireshark
• In the Filter field, type "acdr" to view the debug recording messages
• Note that the source IP address of the messages is by default the OAMP IP address of the
device
• The device adds the header "AUDIOCODES DEBUG RECORDING" to each debug recording
message

437
Viewing DR Messages in Wireshark

ACDR Filter

Proprietary
Header

438
Viewing DR Messages in Wireshark

ACDR and
Q.931 Filter

Proprietary
Header

439
Activating the DR through the WEB Interface

• To set the address/port of the debug recording server:

Defines the IP address of the server Defines the port of the server for capturing
for capturing debug recording debug recording. The default is 925

Defines the threshold (in percentage) for automatically switching to a different debug level, depending on CPU usage
The parameter is applicable only if the 'Syslog CPU Protection' parameter is enabled
440
Logging Filters

• The Logging Filters table lets you configure rules for filtering debug recording
packets, Syslog messages, and Call Detail Records (CDR)
• Example:
• A rule to generate Syslog messages only for calls belonging to IP Groups 2 and 4, or for calls
belonging to all IP Groups except IP Group 3
• Debug recording log filters can include:
• Signaling information (such as SIP messages)
• Syslog messages
• PSTN traces (ISDN and CAS)
• CDRs
• Media (RTP, RTCP, and T.38)
• Pulse-code modulation (PCM) of voice signals from and to the TDM
• Log Filters can be enabled or disabled
441
Configuring filtering rules

• To configure logging filtering rules:

442
Configuring filtering rules

Defines the value for the selected Filtering Type

Defines where the device sends the log file

0. Syslog Server
1. Debug Recording Server (Default)
2. Local Storage
3. Call Flow Server

Defines the filter criteria:

Defines the type of messages to include in the log file
1. Any (default)
0. (Default) Not configured
2. Trunk ID = Filters log by Trunk ID (only Gateway application)
1. Signaling (only Debug Recording)
3. Trunk Group ID = Filters log by Trunk Group ID (only Gateway application)
2. Signaling & Media (only Debug Recording)
4. Trunk & B-channel = (only Gateway application)
3. Signaling & Media & PCM (only Debug Recording)
5. FXS or FXO = (only Gateway application)
4. PSTN Trace (only Debug Recording)
6. Tel-to-IP = Filters log by Tel-to-IP routing rule (only Gateway application)
5. CDR Only (applicable only if the 'Log Destination' parameter is
7. IP-to-Tel = Filters log by IP-to-Tel routing rule (only Gateway application)
configured to Syslog Server or Local Storage
8. IP Group = Filters log by IP Group
9. SRD = Filters log by SRD
10. Classification = Filters log by Classification rule (only SBC application)
11. IP-to-IP Routing = Filters log by IP-to-IP routing rule (only SBC application)
12. User = Filters log by user
13. IP Trace = Filters log by an IP network trace, Wireshark-like expression
14. SIP Interface = Filters log by SIP Interface Enables (default) or disables the rule

443
Configuring generic filtering rules for a SIP call

444
Enabling Traces for an ISDN call

445
Configuring generic filtering rules for a PSTN call

446
Hands-on Lab 11

Debugging Tools
 Thank You

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