[SR-Users] kamailio / rtpproxy - remove codec

[Oliver Roth]

Here we go

IN

INVITE sip:41442742931 at 81.7.235.180:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.197.185.185;branch=z9hG4bKa368.02f63ac6.0
Via: SIP/2.0/UDP 195.216.67.103:5060;branch=z9hG4bK008082590C36C1313E749FCBA4C6
Route: <sip:82.197.185.186;lr=on>
From: <sip:+41446512001 at 195.216.67.103;user=phone>;tag=008082590C36C1313BF6EF12E7F2
To: <sip:+41442742931 at 82.197.185.185;user=phone>;tag=snl_0014532334
Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd at 127.0.0.1
CSeq: 5466 INVITE
Contact: <sip:+41446512001 at 195.216.67.103:5060>
Max-Forwards: 69
Content-Type: application/sdp
Session-Expires: 1800;refresher=uas
Supported: 100rel, timer, replaces
Content-Length:   330

v=0
o=- 82545395 2 IN IP4 195.216.67.103
s=session
t=0 0
m=audio 4550 RTP/AVP 8 0
c=IN IP4 195.216.67.120
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
m=image 4552 udptl t38
c=IN IP4 195.216.67.120
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF





OUT

INVITE sip:41442742931 at 81.7.235.180:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.197.185.186;branch=z9hG4bKa368.aca64b5.0
Via: SIP/2.0/UDP 82.197.185.185;branch=z9hG4bKa368.02f63ac6.0
Via: SIP/2.0/UDP 195.216.67.103:5060;branch=z9hG4bK008082590C36C1313E749FCBA4C6
From: <sip:+41446512001 at 195.216.67.103;user=phone>;tag=008082590C36C1313BF6EF12E7F2
To: <sip:+41442742931 at 82.197.185.185;user=phone>;tag=snl_0014532334
Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd at 127.0.0.1
CSeq: 5466 INVITE
Contact: <sip:+41446512001 at 195.216.67.103:5060>
Max-Forwards: 68
Content-Type: application/sdp
Session-Expires: 1800;refresher=uas
Supported: 100rel, timer, replaces
Content-Length:   188

v=0
o=- 82545395 2 IN IP4 82.197.185.186
s=session
t=0 0
m=audio 64748 RTP/AVP 8 0
c=IN IP4 82.197.185.186
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
5037682.197.185.186

Von: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla
Gesendet: Montag, 7. April 2014 10:40
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] kamailio / rtpproxy - remove codec

Hello,

can you give the incoming sdp and outgoing sdp (both as text) when you are using sdp_remove_media, to see what gets malformed there?

Cheers,
Daniel
On 07/04/14 08:37, Oliver Roth wrote:
Hi all

We use kamailio 3.3.7 and rtpproxy for enduser call-termination.

In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38.
As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488.

How is it possible to remove the whole T38 part of this invite?

We tried
sdp_remove_codecs_by_name(list) without success – what “name” should we use for T38? [T38, t38, t.38, T.38, …]
sdp_remove_line_by_prefix(string)
sdp_remove_media(type)

None of these functions did really work – best was the last one with type=image but then the sip header is malformed.

As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help?

So basically the question is:
How to remove the t38 part of the fax invite? (see attachment)

KR,
Oli

[cid:image001.png at 01CF5259.0D7BFC30]







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Kamailio World Conference - April 2-4, 2014, Berlin, Germany

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