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Signaling Standards
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SIP-T SG MGC SIP SIP User Agent PSTN H.248 Megaco H.323 call signalling PSTN H.248 Megaco MGC SG
H.323 Endpoint MG Call signalling Media gateway control signalling Media flows
For APRICOT, Feb 27, 2001 - 3
MG
H.248/MEGACO Overview
MEdia GAteway Control Protocol [RFC3015] H.248 is ITU-T reference for the same protocol Protocol for controlling telephony gateway and
terminals (IP Phones)
H.248/Megaco evolution
IPDC
MGCP proposal by merging IPDC and SGCP (Telcordia & Level 3)
SGCP
MGCP
I-RFC 2705
MDCP
(proposal) Consensus between IETF and ITU on Megaco Protocol (March 99)
Megaco/H.248
MG
Cn
Tb
O2=I1+I3 I2 O3=I1+I2 I3
Context
Defines communication between Terminations, O1=I2+I3 acts as a mixing bridge Contains 1 or more Terminations Supports multiple streams
Ta
Tc
Stream
A context can have multiple streams, each typically for a medium, e.g. audio, video, etc The MGC specifies which streams a given termination supports
Td
Megaco/H.248: Commands
Provides control for manipulating terminations and contexts. Current Command Set:
Command
Add Modify Move Subtract AuditValue AuditCapabilities Notify ServiceChange
Initiator
MGC MGC MGC MGC MGC MGC MG MGC MG
Description
Adds a termination to a context. Modifies a terminations properties, events, and signals. Moves a termination from one context to another. Removes a termination from its context. Returns current state of properties, events, signals, and statistics. Returns all possible values for termination properties, events, and signals allowed by an MG. Informs MGC of event occurrence(s). Takes or places a termination(s) out of or in service. For registration and restart; notifies MGC termination(s) will be taken out of or returned to service.
Profiles:
Define particular applications of Megaco/H.248 Created by industry organizations, e.g. TIA Specifies which packages are to be supported and other protocol options
18 participants including MGCs, MGs, and Test equipment Ad-hoc group formed to test Megaco/H.248 implementations and collect feedback on specification issues Line to Line and Line to Trunk calls with and without digit collection completed with bearer path
SIP overview
SIP (Session Initiation Protocol IETF RFC 2543) Application-layer signaling protocol for creating,
modifying and terminating sessions with one or more participants
SIP overview
Light-weight generic signaling protocol
Public IP Network
Proxy Server
Registrar
Private IP Network
Redirect Server
Location Server
SIP messages
Message consists of initial line, headers and body There are two types of SIP message
Request Response
First line contains the method being invoked, e.g. INVITE RFC 2543 methods include INVITE, ACK, BYE, REGISTER, CANCEL, OPTIONS First line contains the response code
Headers provide information needed to process or route the message Body contains Session Description Protocol (SDP) describing media
flows or other materials such as encapsulated ISUP messages.
New methods and header types can be added at any time without
changing the protocol
For APRICOT, Feb 27, 2001 - 14
SIP status
Base protocol is RFC2543
Presently being revised RFC2543bis is under discussion
SIP-T
SIP-T = Session Initiation Protocol Telephony
previously referred to as SIP+ or SIP BCP-T a collection of internet drafts that extend SIP to support inter-Media Gateway Controller (MGC) communications. SIP-T is an interface agreement on a collection of standards as opposed to a separate protocol
SIP-T describes how to interwork SIP and ISUP SIP-T directly negotiates a media connection between
gateways. Endpoint information is carried in SDP (Session Description Protocol) which can describe both IP and ATM endpoints.
SIP-T purpose
Proxy
IP network
SS7 network
Proxy
Proxy
SS7 network
PSTN to IP interworking
MGC
SIP agent
IP network
SS7 network
Proxy
SIP-T Implementation
Three major extensions required:
INFO method extension to the base protocol in addition to others
SIPT status
Work on SIP-T was initiated by the International
SoftSwitch Consortium (ISC) in early 1999.
H.323 Overview
Packet-based multi-media communications system It includes several protocols
H.225.0 RAS (registration, admission, status ) H.225.0 Call Signalling H.245 Logical channel signalling and media control RFC 1889 RTP/RTCP for media transport H.450.x Supplementary services H.225.0 Annex G Inter-domain registration and billing information exchange
H.261 H.263
T.120
Control
Gatekeeper
address translation (IP, telephone) admission control cannot generate or terminate calls
Endpoints
MCU
Gateway
can make or receive calls Realized by terminals and logically present in Gateways
H.245 and H.225.0 also updated Currently working on version 5. No release date
specified.
SIP
IETF SIP, MMusic, ... Modular, simplistic design Easily extended & updated Based on Web principals (Internetfriendly) Readily extensible beyond telephony
No real end-device features std, yet Many options for advanced telephony features (need to make specific choices) Astounding progress, velocity Rapidly growing industry momentum, at system and device level Growing interest in SIP-Phones and soft clients, products appearing
SIP is anticipated long-term winner, but H.323 networks will remain for some time, and are a source of revenue in the international market
For APRICOT, Feb 27, 2001 - 25
http://www.ietf.org/
ITU:
http://www.itu.int/ITU-T/index.html
Most Megaco/H.248 related documents should also be available through Megaco WG, above
Thank You!