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SIP on an IMG

Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call whereby an IMG can be used as a Media Gateway to allow two separate networks to connect. The IMG supports SS7 to SIP, ISDN to SIP, CAS to SIP, SIP to SIP, and H.323 to SIP. Below is exemplary diagram of an IMG in a TDM to IP network.

 

Supported SIP Features

RFC

Description

2246

Transport Layer Security (TLS) for SIP

2327

Session Description Protocol (SDP)

2976

SIP Info

3240

Internet media type message/sipfrag

3261

SIP: Session Initiation Protocol

3262

SIP PRACK

3263

Locating SIP servers for DNS lookup SRV and A records

3264

SDP Offer/Answer Model (Do not support multiple 'm' lines in SIP SDP)

3265

SIP Subscribe/Notify

3311

SIP Update

3323

SIP Privacy Header

3325

Asserted Identity

3326

SIP Reason Header

3332

M3UA Adaption Layer

3372

SIP for Telephones (SIP-T/SIP-I)

3398

ISUP/SIP Mapping

3515

SIP Refer

3551

Payload Type Support

3578

ISUP Overlap Signaling to SIP

3581

Symmetric Response Routing

3666

Call Flows - SIP to PSTN Dialing

3711

IP Media Layer Security Standard (RTP/RTCP)

3725

Third Party Call Control for SIP

3764

ENUM for SIP Address of Record

3891

SIP Replace Header

3892

SIP Referred by Mechanism

4028

SIP Session Timer

4040

Clear Channel Codec Support

4244

SIP History info (for call diversion)

4568

IP Signaling Layer Security Standard (RTP/RTCP)

4904

Trunk Group Parameter Support

Basic Support

  • RFC 3261, SIP (Session Initiation Protocol)
  • Backward compatible with entities running RFC 2543
  • RFC 3581, Rport Extension Parameter in the Via Header
  • RFC 2327 SDP Support
  • RFC 3551 RTP Profile for Audio and Video Conferees with minimal control.
  • RFC 3666 Call Flows -- SIP to PSTN Dialing
  • RFC 3960 - Early Media and Ringing Tone Generation
  • Transmission Control Protocol Support (TCP/IP). Single or multi-socket use.
  • Reliable User Datagram Protocol (UDP) transport, with retransmissions
  • SIP Authentication and Outbound Registration. The IMG does not support inbound registration. Inbound Registration is not applicable with Media Gateways. For more information click on IMG 1010 - Authentication and Outbound Registration Link
  • Call Release Data in the Radius CDR (Software 10.5.0 +) Click IMG 1010 - Call Release Origin in CDR for more information
  • Vocoder Data in the Radius CDR (Software 10.5.0 +). Click IMG 1010 - RADIUS - Codec Info in CDR for more information
  • The IMG supports being a User Agent Client (UAC) or User Agent Server (UAS) and will inter-operate with SIP proxies.
  • Supports Early Media. Supports 180/183 Session Progress. Click IMG 1010 - Early Media for more information.
  • Can create multiple SIP profiles. A SIP Profile allows you to easily assign a number of SIP features to a Physical IMG. You create a SIP Profile and then assign profiles to a gateway in the External Gateway pane. See IMG 1010 - SIP Profiles link.
  • Supported Response Messages 1xx, 2xx, 3xx, 4xx, 5xx, 6xx
  • SIP Session Timer. Click on IMG 1010 - SIP Session Timer link for more information.
  • SIP Redirection. Click on IMG 1010 - SIP Redirect Server Support link for more information
  • SIP Transcoding. Click on IMG 1010 - Transcoding link for more information.
  • SIP Call Hold. Click on IMG 1010 - SIP Call Hold link for more information.

Supported Methods

SIP Extensions

Routing/Call Handling

Media

  • Network Address Translation (NAT) Traversal. See IMG 1010 - Symmetric NAT Traversal for more information
  • Relay for Dual Tone Multi Frequency (DTMF) digits, including payload type negotiation (RFC 2833)
  • Codec Negotiation Priority. See IMG 1010 - SIP Codec Negotiation Priority Selection for more information
  • T.38 Fax support. See IMG 1010 - SIP Fax-Modem for more information.
  • Support for RFC 3550 (RTP: A Transport Protocol for Real-Time Applications) – Partially compliant.

    If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message, the gateway returns the same rate in the response SDP.

Interworking

Basic Support:

SIP

  • SIP 3xx Gateway Responses
  • SIP Diversion Header
  • SIP Trunk Group ID's
  • SIP Codec Negotiation
  • SIP Busy Out Modem BypasS
  • SIP-T
  • SIP-I
  • SIP over TLS
  • SIP Refer
  • SIP Refer for Call Transfer
  • SIP Dial Around Indicator Support
  • Generic Name Indicator (SIP to SS7)
  • M3UA Signaling Gateway for TCAP/SCCP
     
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