Here we go

 

IN

 

INVITE sip:41442742931@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@195.216.67.103;user=phone>;tag=008082590C36C1313BF6EF12E7F2

To: <sip:+41442742931@82.197.185.185;user=phone>;tag=snl_0014532334

Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd@127.0.0.1

CSeq: 5466 INVITE

Contact: <sip:+41446512001@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@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@195.216.67.103;user=phone>;tag=008082590C36C1313BF6EF12E7F2

To: <sip:+41442742931@82.197.185.185;user=phone>;tag=snl_0014532334

Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd@127.0.0.1

CSeq: 5466 INVITE

Contact: <sip:+41446512001@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@lists.sip-router.org [mailto:sr-users-bounces@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

 

 

 

 




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