Signalling and Control In IP Networks - H.248, H.323 and SIP

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Signalling and Control In IP Networks - H.248,
H.323 and SIP

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Signaling Standards

• Media Gateway Control Signalling
• H.248/Megaco
• Call Signalling
• SIP and SIP-T
• H.323

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Network View
SIP-T
SG
MGC
SG
MGC
SIP
SIP User Agent
H.323 call signalling
H.248 Megaco
H.248 Megaco
H.323 Endpoint
MG
MG
Call signalling
Media gateway control signalling
Media flows
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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)
• Basis for Vendor Independent Network deployment

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H.248/Megaco evolution
MGCP proposal by merging IPDC and
SGCP (Telcordia Level 3)
IPDC
MGCP released as Informational RFC (Oct 99)
MGCP
I-RFC 2705
SGCP
Lucent submits MDCP to ITU-T SG16 (Nov 1999)
MDCP (proposal)
Consensus between IETF and ITU on Megaco Protocol
(March 99)
IETF RFC 3015
Megaco/H.248
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Megaco architecture

• Media Gateway Control Layer (MGC)
• Contains all call control intelligence

• Media Gateway Control Protocol
• Master / slave control of MGs by MGCs
• Connection control
• Device control and configuration
• Events and signals
• Orthogonal to call control protocols

• Media Gateway Layer (MG)
• Implements connections
• Implements or controls end device features
  (including UI)
• No knowledge of call level features

Analog Media Gateway
IP Phone Media Gateway
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Megaco connection model

• Based on 3 concepts
• Termination
• Identifies an end point for media flows
• Implements Signals, and generates Events
• Can appear in at most one context.
• Permanent (provisioned) terminations can exist
  outside a context
• Context
• Defines communication between Terminations, acts
  as a mixing bridge
• Contains 1 or more Terminations
• Supports multiple streams
• 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

O1I2I3
Simple, powerful connection/resource model
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Megaco/H.248 Commands

• Provides control for manipulating terminations
  and contexts.
• Current Command Set

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Packages and profiles

• Packages
• Extension mechanism to define new termination and
  MG behavior
• Adds detailed application semantics to the
  protocol, e.g. a package can define the events
  for DTMF signaling
• 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

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Megaco/H.248 Standards Status

• Standardization of Base Protocol DONE
• Decided as ITU-T Recommendation H.248 in June
  2000
• Published as IETF Megaco RFC 3015, in Nov 2000
• Continued work on bug fixes for the base protocol
• First issue of H.248 Implementors Guide
  approved, Nov 2000
• Package definition Rapid Progression
• Starter Kit (Annex E) defined in base protocol
• Additional packages defined as Annexes F, G, J
  and K to H.248 in Nov 2000
• More ongoing in both IETF and ITU
• Megaco/H.248 Interoperability Second Event Just
  Completed!
• First multi-vendor interoperability event was
  great success (Aug 2000)
• 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
• Second event just occurred last week (Feb 2001)

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SIP overview

• SIP (Session Initiation Protocol IETF RFC 2543)
• Application-layer signaling protocol for
  creating, modifying and terminating sessions with
  one or more participants

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SIP overview

• Light-weight generic signaling protocol
• Used to initiate sessions and invite members to a
  session
• Text-based protocol (good for prototyping)
• Syntax is textual and based on HTTP
• There have been several bake-offs with different
  vendors demonstrating interoperability of basic
  calls

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SIP Architectural Model
Proxy Server
Registrar
SIP Agent
UAC
Redirect Server
SIP Agent
UAS
Location Server
UAC
UAS
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SIP messages

• Message consists of initial line, headers and
  body
• There are two types of SIP message
• Request
• Response
• Requests are always initiated by a UAC function
• First line contains the method being invoked,
  e.g. INVITE
• RFC 2543 methods include INVITE, ACK, BYE,
  REGISTER, CANCEL, OPTIONS
• Responses are generated by servers
• 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

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SIP status

• Base protocol is RFC2543
• Presently being revised RFC2543bis is under
  discussion
• Standardized in the IETF as RFC 2543 in March 99
  (now being further refined in the SIP working
  group).
• RFC 2543 just covers basic functionality. There
  are several related internet drafts covering
  services.
• Has rapidly growing industry momentum
• Intense efforts underway to develop
  service-specific extensions

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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.

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SIP-T purpose
PSTN bridging (PSTN - IP - PSTN) PSTN Signaling
is carried transparently over the IP network
Proxy
SIP enabled network
PSTN to IP interworking
SIPagent
MGC
IP network
Proxy
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SIP-T Technical Approaches

• SIP-T uses two approaches
• Map ISUP message contents to fields in the SIP
  header for interworking with pure SIP agents
• Encapsulate ISUP message within SIP message body
  for PSTN bridging

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SIP-T Implementation

• Three major extensions required
• INFO method extension to the base protocol in
  addition to others
• Session initiation and teardown is not enough
• Mid-Call events
• More complex services enabled
• MIME Type addition
• Standard method of encapsulating legacy signaling
• Simplifies Inter-working
• Local variant
• Interconnect variant (LCD)
• ISUP to SIP Mapping

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SIPT status

• Work on SIP-T was initiated by the International
  SoftSwitch Consortium (ISC) in early 1999.
• SIP-T is still a work in progress. The ISC is
  continuing to develop profiles for SIP and
  telephony interworking.

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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
• The original VoIP protocol suite
• Whole System Architecture
• Provides Interoperability
• Transport independence
• Platform and application independence
• Multipoint support
• Primarily used in corporate networks

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H.323 overview continued

• Comprised of several protocols

System Control User Interface
Data
Video
Audio
H.245 Control
H.225
T.120
H.261 H.263
G.711 G.722 G.723 G.728 G.729
Call Control
RAS
RTP/RTCP
AAL5
UDP or TCP
UDP
ATM
IP
Lower Layers Vary
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H.323 elements and entities

• Terminals
• PCs, IP phones, set-top boxes
• Audio
• Video (optional)
• Data (optional)

• Gatekeeper
• address translation (IP, telephone)
• admission control
• cannot generate or terminate calls

• Endpoints
• can make or receive calls
• Realized by terminals and logically present in
  Gateways

• Gateway
• Interworking with
• other multimedia terminals
• GSTN

• Multipoint Control Unit (MCU)
• Support for multipoint conferences
• Always contains a MC
• Optionally contains an MP

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H.323 current status

• H.323 version 4
• Includes many major changes
• A significant number of contributions from Nortel
  Networks
• Approved in November 2000
• H.245 and H.225.0 also updated
• Currently working on version 5. No release date
  specified.

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H.323/SIP comparison summary
H.323 SIP
Stds Body ITU-T SG-16 IETF SIP, MMusic, ...
Properties Complex, monolithic design Difficult to extend update Based on H.320 conferencing and ISDN Q.931 legacy (Bell headed) Limited potential beyond telephony Some QoS built in CODEC types specified Higher degree of interoperability Modular, simplistic design Easily extended updated Based on Web principals (Internet-friendly) Readily extensible beyond telephony
Status w.r.t. end device H.450.x series provides minimal feature set only, pure peer approach Adding ( mixed peer/stimulus approach soon poor architecture) Slow moving No real end-device features std, yet Many options for advanced telephony features (need to make specific choices) Astounding progress, velocity
Industry acceptance Established now, primarily system level Few if any H.323-base telephones End-user primarily driven by Siemens, Microsoft (NetMeeting), Intel 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
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For More Information

• IETF
• IETF home page
• http//www.ietf.org/
• Internet-draft search engine
• http//search.ietf.org/search/brokers/internet-dra
  fts/query.html
• RFC search engine
• http//www.rfc-editor.org/rfcsearch.html
• Megaco WG charter
• http//ietf.org/html.charters/megaco-charter.html
• Megaco documents repository
• ftp//standards.nortelnetworks.com/megaco/
• ftp//standards.nortelnetworks.com/megaco/docs/lat
  est/
• ITU
• ITU home page
• http//www.itu.int/ITU-T/index.html
• SG-16 document repository (H.323, H.248)
• ftp//standard.pictel.com/avc-site

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Thank You!