[SIPForum-discussion] Question About media attribute within SDP

Shakil, Rashid rshakil at covad.com
Mon Jul 27 14:16:59 UTC 2009


Thanks for the detailed reply. Your analysis is correct even though I removed the rtpmap from SDP my fax calls are still failing.

From: Garron, James [mailto:jgarron at sonusnet.com]
Sent: Monday, July 27, 2009 10:11 AM
To: Shakil, Rashid; discussion at sipforum.org
Subject: RE: [SIPForum-discussion] Question About media attribute within SDP

Rashid, assuming that the devices in the flow follow draft-ietf-sipping-realtimefax-00/01 then I do not think that this rtpmap should be in issue.  Upon fax detection there should be a re-INVITE to pass-through mode (or T.38) that would remove this.

4.4 Internet telephony gateways and fax pass-through mode

   This section deals real-time facsimile communications over IP using
   fax pass-through instead of T.38 fax relay.  In this mode, the
   facsimile communication is handled as a PCM audio call (PCMA/PCMU as
   specified in ITU-T recommendation G.711).

   The fax pass-through mode is important to prevent call failures, for
   example in cases when one of the SIP communication peers does not
   support T.38.  For Internet telephony gateways with support for
   PCM/G.711 audio but no support for T.38 real-time fax, it is
   recommended to switch the session to fax pass-through mode.
   Internet telephony gateways SHOULD handle the fall back mode to fax
   pass-through by recognizing the SDP T.38 connection and proposing to
   switch to a new audio connection.  The new audio connection SHOULD
   have the following characteristics: at a minimum, specify PCM G.711
   codec, silence suppression OFF. The telephony gateway that
   originated the failed T.38 re-INVITE SHOULD initiate the subsequent
   re-INVITE to fax pass-through mode.
   Refer to section 6.2 for a sample call flow illustrating this
   scenario.

________________________________
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shakil, Rashid
Sent: Friday, July 24, 2009 10:21 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Question About media attribute within SDP

Hello All,

For outbound calls the customer's Cisco IAD 2431 generating following SDP. The SIP peer doesn't like the media parameter "a=rtpmap:19 CN/8000" in following SDP.  Two quick questions,


1.       What "a=rtpmap:19 CN/8000" represent in this SDP

2.       How can it be removed

3.       Is it possible that it cause fax failures ?


v=0
o=- 2847654225 2847654225 IN IP4 72.245.184.130
s=SIP Call
c=IN IP4 67.102.244.5
t=0 0
m=audio 17746 RTP/AVP 0 101 19
c=IN IP4 67.102.244.5
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

Thanks in advance for the input

Regards,

Rashid Shakil



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