[Freeswitch-users] rtpmap line missing on answer

[Sean Eichhorn]

Yes, I understand that this is the Cisco proprietary codec.  The problem is that between two Cisco systems the SIP endpoints will not use this protocol if it is not sent in the SDP.  The *original* SDP does contain the codec, but when the call is answered, another SDP is sent.  This SDP is altered by Freeswitch and all codecs other than the selected codec, NTE, and comfort noise are removed.
It might not be Freeswitch doing it.  I’m digging into the source and it looks like it might be the Sofia-SIP library.
The funny thing is that it works in one direction, but not the other.
FS subscriber -> CIDR defined static SIP endpoint (WORKS!)
CIDR defined SIP endpoint -> FS Subscriber (Doesn’t work)

It seems that one of the major differences is that the subscriber answers with a 200 OK, whereas the CIDR endpoint answers with a 183.

I’m currently searching to see if there is a way to change the response type, but so far no luck.

Sean Eichhorn
General Communications Inc.
IP Telephony Engineer
(907) 868-6902

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From: David Ponzone [mailto:david.ponzone at ipeva.fr]
Sent: Friday, May 20, 2011 01:03
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtpmap line missing on answer

Sean,

X-NSE is the Clear Channel codec from Cisco.
I suspect FreeSWITCH does not support it, and as such, it can't be offered to leg B.
Perhaps there is a way to enable in outbound codecs as bypass, but I really doubt so.
Though, you could try to enable late-negotiation where client and gateway will negotiate together.

David Ponzone  Direction Technique
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Le 20/05/2011 à 20:54, Sean Eichhorn a écrit :


I have a freeswitch system that (for the most part) works flawlessly for me.  However, I need to have an additional rtpmap line in the SDP when a call is answered.
My client sends the following SDP upon answering the call :
   m=audio 17214 RTP/AVP 0 19 101 100
   c=IN IP4 192.168.98.79
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:100 X-NSE/8000
   a=fmtp:100 192-194

Freeswitch forwards the following SDP to the external gateway :
   m=audio 20654 RTP/AVP 0 19 101
   c=IN IP4 66.223.187.208
   a=rtpmap:0 PCMU/8000
   a=rtpmap:19 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16

The other information appears fine.  Everything works, but my client-side is not receiving the a=rtpmap:100 line.
This issue appears regardless of whether or not I’m in proxy mode, bypass mode, or neither.

Using "sip_append_audio_sdp” has no effect on the answering SDP, only the initial offer.

Any ideas?

Thanks in advance,
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