Introduction to
Voice Over IP (VOIP) Protocols
By J K Ramachandra
Background
Currently, two Independent networks (voice and
data)
Circuit Switched and packet switched
Both undergone tremendous enhancements over
the years
Packet network is relatively a later entry, but more
aggressive
Convergence combining both into one network
talked quite often from early eighties
ISDN/SS7 was the first instance of convergence
Voice network
Fixed (assured) bandwidth, optimally matched for
Poisson voice traffic
High reliability
Five 9s Availability (5 min/ year downtime)
Disadvantages:
Inefficient usage of bandwidth
Closed architecture, single vendor
Long turn-around time for new features (18 months)
Data network
Rapid growth of data traffic
Data traffic has overtaken US voice traffic in 1998, UK
in 1999 (voice 8% total US traffic by 2004)
US Internet users increased from 10 to 50 million in 4
years
Truly global infrastructure with minimum barriers
Disadvantage:
best effort and not suitable for real time
Not designed for voice traffic (short duration, high call
rate)
Need for VOIP protocols
Use the advantages of IP
QOS still to be addressed
Still large PSTN network base
User the POTS the Black phone - continue to
be used
Co-existence of both networks for quite some time
Address multi-media (audio, video, data) session
instead of just plain voice calls
Protocols defined for complete IP based as well as
inter-working solutions
VOIP protocols
Media Gateway Control Protocols
MGCP, MEGACO (H.248)
H.323
the first VOIP protocol defined by ITU-T
SIGTRAN suite of protocols
To carry signaling (ISDN/SS7) over IP
Session Initiation Protocol (SIP)
Text based (like HTTP); simple and extendable; Likely
to be the protocol of the future
Real Time Transmission Protocol (RTP)
for voice (media) transmission
Basic Switching functions
Call setup
Call Origination (off-hook)
Feed dial tone
Collect digits
Translate digits and route selection
Terminate the call
Called alert and ring back
Answer and connect
Conversation
Speech transmission between users
Call Tear Down
Disconnect from one of the user
Disconnect both parties
SIGNALLING
Media transmission
SIGNALLING
Conventional SWITCH MODEL
Control
SERVICES/
FEATURES
SIGNALLING
voice
Signaling
CALL
CONTROL
Users
User
Access
SWITCH/Matrix
RESOURCES
Co-located (Normally), single integrated system
PSTN/SCN (Switched Circuit Network)
Next Generation Network SWITCH
MODEL
CALL
SIGNALLING
CONTROL
SERVICES/
FEATURES
IP
User
Access
RESOURCES
Physically decomposed, distributed Architecture
Proposed by ETSI (TIPHON) - MG, SG, MGC
Standards generated by ITU-T, IETF
NGN ELEMENTS
SIGNALLING
GATEWAY
FEATURE
SERVER
Call Agent/MGC
Announcement Server
MEDIA
GATEWAY
IP
CO/IXC
Trunks
H.323
MEDIA
GATEWAY
Trunking
Gateway
SIP
MEDIA
GATEWAY
POTS ISDN
RESIDENTIAL/ACCESS GATEWAY
Media Refers to traffic (Voice, video, data)
Media Gateway Media converter between different
networks (SCN to IP)
Media Gateway Control Protocols
MEGACO (RFC 3015)
MGCP (RFC2705)
Media Gateways (MG)
An interface between PSTN and IP network
Converts and passes Speech samples between the
two networks
PCM on PSTN === RTP (different codecs) on IP
Communicates with MGC for call handling using
MEGACO/MGCP
Responsible for feeding tones/digit collection/off
hook/on hook detection, voice compression etc..
Highly hardware intensive (DSP, embedded
systems)
Media Gateways Controllers (MGC)
Also called as Call agent or Soft switch
Call controlling agent for next generation
Controls MGs using MEGACO/MGCP
Highly software intensive and mostly uses
standard computing hardware platform like
sun servers
Gateway Control Protocols
RTP/RTCP
RTP/RTCP
IP Network
MG
MG
MGCP/MEGACO
Application
MEGACO/MGCP
TCP/UDP
IP
Data Link
Physical
MGCP/MEGACO
MGC
Application
RTP/RTCP
UDP
IP
Data Link
Physical
Types of Media Gateways
Access Gateway
ISDN user to Network Interface
E.g. ISDN PBX
Residential gateway
Analogue Lines (RJ11) (typically One or two)
Trunking Gateway
SCN to IP Network, T1/E1 lines,
Normally uses SS7 for signalling
Network Deployment Options
SCP
STP
LEC
PBX
IXC
LEC
ACC GW
TRK GW
MGC
TRK GW
RES GW
IP Network
MEGACO
Assumptions:
Call Processing intelligence with MGC
MGs are as dumb as possible
MGC instructs MG for every possible stage of
the call
MEGACO-Connection model
PSTN
Network
DS0 Ch
Physical
Context C1
RTP Stream,
Ephemeral
IP
Network
Connection model consists of Terminations and context
Physical terminations - equivalent to circuit#
Ephemeral termination - RTP streams, exists only for the
duration of their usage
Context is an association between a set of terminations
Context is created and maintained by MG
A termination exist in one context at a time
Packages
Package identifies the type of termination
Each Termination is associated with a
package:
e.g. Analog Line,
DS0,
Tone Gen.,
DTMF Gen.,
DTMF detection ,
etc..
Signals
MGC orders MG to apply a signal
Signals are applied to terminations
e.g. Dial tone, Ring, DTMF tones (O/G side)
Can apply Set of signals in a sequence
Signal Type
on/off (e.g. Announcements)
Timeout (e.g. dial tone, ring)
Brief (e.g. DTMF tone)
Events
MGC orders MG to detect events
Events are detect by MG on termination
e.g. on/off hook, DTMF digits
MG notifies the requested events after detection
List of events can be requested
Embedding with another Event/Signal
e.g detect Off hook event and apply dial tone signal
Digit Map
Simple way to instruct MG to collect sequence of
digits as per dial plan
(0S|00S|[1-7]xLxx|8Lxxxxxxx|*xx|9L1xxx)
S short timer
L Long timer
x any digit
| or
[1-7] Any digit between 1 to 7
MG will look for the best match and report the
digits to MGC
Media Negotiation
Different types of codecs can be used for the call
Session Description Protocol (SDP) is used for
codec negotiation
Originator (MG) sends set of codec choices
MGC passes them to terminator
Terminator normally selects one of them and sends it
back
MGC sends the selected one to originator
Thus, selected codec will be used for the call
Typical codecs:
G.711 (u/a law, 64 KBPS), G.729 (8 KBPS). G.723.1 (5.3/6/3
KBPS)
MEGACO/MGCP-Commands
ADD (MGC to MG)
Add terminations to a context.
Add on first terminations creates Context.
Create connection (CRCX) in MGCP
Modify (MGC to MG)
Modifies the properties, events and signals of a
termination
Move (MGC to MG)
Moves a termination to another context
Commands- Continued
Subtract (MGC to MG)
Disconnects a termination from a context
Returns Statistics of that termination
If last termination, deletes the context
Delete connection (DLCX) in MGCP
Notify (MG to MGC)
MG to inform the occurrence of the requested
events
Commands- continued
Audit Value (MGC to MG)
Current state of properties, events and signals
of a termination
Audit Capabilities (MGC to MG)
All the possible values for the termination
properties, events, signals allowed by MG.
Commands - continued
Service Change (mainly from MG to MGC)
Need for REGISTRATION due to physically
distributed architecture
Registration with MGC (primary and list of secondary
MGCs)
Indicate termination state (OOS/INS) by MG
Hand over to another MGC by MGC
Request MG to take termination OOS (MEGACO
only).
Restart In Progress (RSIP) in MGCP
MEGACO-transactions
Transaction Request: Transaction ID1
Context1 (Command 11, Command 12, Command 13
Context2 (Command 21, Command 22)
Transaction Reply: Transaction ID1
Context1 (Response 11, Response 12, Response 13
Context2 (Response 21, Response 22)
MEGACO message -Example
MEGACO/1 [130.202.9.126]
Transaction = 10003 {
Context = - {
Modify = ds31/ds13/ds05 {
Events = 2223 {
al/on, dd/ce {DigitMap=DialPlan0}
Signals {cg/dt},
DigitMap = DialPlan0 {
(os|00s||8Lxxxx)}
}
}
}
---------------------------------------------------------------------------MEGACO/1 [130.202.9.121]:55555
Reply = 10003 {
Context = - {Modify = ds31/ds13/ds05 }
}
MGCP-Example
CRCX 24323 [email protected] MGCP 1.0
C: A3C47F223456789F0
K: 6234-6237, 6888, 19045-19048
M: sendonly
N: [email protected]:5234
L: p:10, a:PCMU, b:32-65, e:off
X: 0123456789B1
R: hd(E(R([0-9#T] (D),hu[N],s[dl],D[0-9].[#T])))
D: (0T| 00T| [1-7]xxx| 8xxxxxxx|#xxxxxxx)
------------------------------------------------------200 24323 OK
I: 234ABC432432
v=0
c=IN IP4 128.96.41.1
M=audio 3456 RTP/AVP 0 96
some highlights of the MGCP/MEGACO
Transaction 3 way handshake:
Transaction Request, Reply/Pending and
Response Ack.
Text/Binary format (only Text in MGCP)
TCP/UDP (only UDP in MGCP)
Retransmission timers, Long timer,
Suspicion/disconnect thresholds
Interim AH security (only in MEGACO)
Call flow-Registration
MG1
MGC
MG2
Service Change
ROOT term, Restart method
Service Change
ROOT term, Restart method
Reply
Self/other MGC Id to try
Reply
Self/other MGC Id to try
Call flow ResGW - origination
MG1
MGC
Modify, termId *,
Event=Offhook
Reply
Notify, termId x1
Observed Event=Offhook, time
Reply
MG2
Call flow-Digit collection
MG1
MGC
Modify, termId x1,
Event=dgt, onhook ,Digit map,
Reply
Notify, termId x1,
Observed Event=Digits, time
Reply
MG2
Call flow-Create Contexts
MG1
MGC
Add, Context $
ADD =termID x1, ADD =$ (ephemeral)
Media=Preferred SDP
Reply, Context C1
Add, Context $
termID x1, termID x2 (ephemeral)
ADD =$, ADD= $(Ephemeral)
Media=Selected SDP choices
Media=MG1 SDP choices, send/recv
Mode=Receive only
Reply, Context C2
termID y1, termID y2 (ephemeral)
Media=selected SDP
MG2
Call flow- Wait for Answer
MG1
MGC
Modify, Context C1
Modify, Context C2
Signals =Ring back
Media=MG2 selected SDP
Reply, Context C1
Signals=Ring, Event=OffHook
Reply, Context C2
Notify, Context C2
Modify, Context C1
Signals =Stop Tone, Event=OnHook
Mode=Send/Receive
Observed Event=OffHook (Answer)
Reply, Context C2
MG2
Call flow-disconnect
MG1
MGC
MG2
Notify, Context C1
Observed Event=onHook
Reply, Context C1
Subtract, Context C1
Audit statistics, termId x1, x2,,
Event=Offhook
Reply, Context C1
Statistics for termId x1, x2
Subtract, Context C2
Audit statistics, termId y1, y2,
Event=Offhook
Reply, Context C2
Statistics for termId y1, y2
H.323 Protocol
H.323 introduction
Standard for multimedia communication
over IP network
For Audio, video and data communications
Umbrella standard -pointer to many other
standards
H.323 Network
PSTN
H.323
Gateway
IP Network
IP/TCP/UDP/RTP/RTCP
Gate Keeper
H.323 Terminal
Multipoint Control Unit
H.323 Gateway
PSTN
H.323 terminal
Client endpoint on the network
Supports:
Audio (G.711, G.722, G.723, G.728, G.729)
Video (H.261, H.263)
Data terminal - T.120
H.225 (Subset of Q.931) Control functions
H.245, Negotiation & media channel setup
RTP -media packets
H.323 Gateways
Provides translation function between
H.323 and Circuit Switched network
Gateway are optional in H.323 network
H.323 Multipoint Control Unit
Supports conferencing between endpoints
H.323 Gate Keeper
Required Functions:
Admission control
Bandwidth control and management
Address translation (E.164 # to IP Address)
Registration/Admission/Status (RAS)
Manages all components of H.323
Optional, but must be consulted if present
H.323 Gate Keeper
Optional Functions:
Call Control Signalling
Call management (call screening, forwarding,
redirection, call routing etc..)
H.323 Protocol Stack
Audio
Call Control
Video
Data
A/V
Control
Control
RTCP
RAS
G.7xx H.26x
H.225 H.245 T.120
(Q.931)
RTP
TCP
UDP
IP
H.323 Call Stages
Discovery and Registration (Who I am)
Call Setup (whom to call)
Call Negotiation (Our Capabilities)
Media Channel Setup (Open audio channel)
Media transport (Send/Receive Audio)
Call Termination (call ends)
Discovery and Registration
Gate Keeper Request GRQ, Who is my GK?
Gate Keeper Confirm/Reject (GCF/GRJ), I am
H.323
Terminal
Registration Request, RRQ, Associate PSTN/IP #
Registration Confirm/Reject (GCF/GRJ), OK
Gate
Keeper
H.323 Call setup
Location Request LRQ, Where is NPANXX-XXXX?
Location Confirm LCF, At this IP Address
H.323
Terminal
Admission Request ARQ, May I call this IP Addr
Admission Confirm, ACF, Yes & use xx KBPS
Gate
Keeper
Call Setup - H.225 (Q.931)
ARQ
ACF
SETUP
Call Proceeding
ARQ
ACF
Alerting
Connect
Media Channel Setup - H.245
Open Logical Channel
Open Logical Channel Ack
Open Logical Channel
Open Logical Channel Ack
Logical Chl opened for each media stream
Logical Chl 0 - for H.245 Control
Media flows now. Possible Bandwidth negotiation during call
Call termination
End Session Command (H.245)
End Session Command
Release Complete (Q.931)
Disconnect Request (RAS)
Disconnect Confirm
Disconnect Request
Disconnect Confirm
Session Initiation Protocol
(SIP)
SIP introduction
SIP is an application layer control protocol that can
establish, modify and terminate multimedia
sessions or calls.
Direct competition with H.323. Microsoft migrated
from net-meeting (H.323) to MSN
messenger/windows XP(SIP)
Mainly addresses all IP network (end user on IP),
though gateways concept are used in real
implementation
SIP message format is based on the Hyper Text
Transport Protocol (HTTP) message format, which
uses a human-readable, text-based encoding.
SIP introduction..contd.
SIP is transport-layer independent protocol i.e. it can
be used with any datagram or stream protocol like
UDP, TCP, SCTP, ATM etc.
It uses the session description protocol (SDP) for
media negotiations (similar to MEGACO/MGCP)
Simple, Extendable and most promising
SIP is also the the most preferred protocol choice for
communication between MGCs (Call Agents).
Accepted as the core signaling protocol for 3rd
generation (3g) wireless
SIP Architecture
Consists of User Agent, Network server (proxy, redirect,
registrar and location server)
SIP User Agent
End system acting on behalf the user
User Agent Client (originator) and User Agent Server
(terminator)
Maintains call state
SIP Registrar
Registrar receives Registration from clients (UA)
regarding current location and stores in a location server
(non-SIP)
This information is used for routing (terminating) the call
SIP Proxy Server
Forwards client requests to another proxy or UA
Either maintain call state (stateful) or can be stateless
Can fork request to multiple servers (parallel search)
Softswitch/Gatekeeper function
SIP Redirect Server
Returns the next address to originator instead of
forwarding
Originator retries with the new address
SIP Messages
Request and Response messages
Request Messages:
REGISTER For registration
INVITE Session initiation
ACK Acknowledge Invite Response
BYE Successful session termination
CANCEL- Terminate the pending request
SIP Messages..Contd.
Response Messages:
1xx: information (100 trying, 180 ringing)
2xx: Successful (200 OK)
3xx: Redirection (301 Moved permanently )
4xx: Client error (400 Bad Request, 404 Not found)
5xx: Server error (503 Service unavailable)
6xx: Global error (603 Decline)
SIP Call flow
User A
Proxy 1
INVITE
100 Trying
Proxy 2
INVITE
100 Trying
180 Ringing
180 Ringing
User B
INVITE
180 Ringing
Alerting User B
200 OK
User B accepts call
200 OK
200 OK
ACK
ACK
ACK
established
Both Way RTP Media
BYE
BYE
200 OK
INVITE
BYE
200 OK
Media (voice) path
200 OK
Called party
hangs up
up first
Conclusion:
MGCP, MEGACO, H.323 or SIP
different ways to establish communication between
the users over IP
