[asterisk-users] Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i

[Gareth Owen]

The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html).  I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again.

If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec until the next beta comes out.


Gareth

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Gareth Owen
> Sent: 06 October, 2006 7:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Codes negotiation
> problemsbetweenAsterisk1.4beta2 and Aastra 480i
> 
> The bad news is that the 1.4.1 beta firmware won't help solve your
> problem, the problem is being caused by the multiple "ptime" directives in
> the INVITE message.
> 
> According to RFC2327 "ptime" is a media-level description and hence
> applies to all the codecs in the "m=audio" line, thus it is only valid to
> have one of these per stream.  Because of this the phones parser is
> rejecting the SDP as being invalid and thus sending back a 488.
> 
> 
> I believe this new functionality has been added by the "RTP Packetization"
> work in 1.4 (see http://bugs.digium.com/view.php?id=5162)
> 
> I'm going to raise a bug against asterisk on this, but at the same time
> I'll try and find a workaround on the phone-side.
> 
> 
> Gareth
> 
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Morten Isaksen
> Sent: 06 October, 2006 10:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Codes negotiation problems
> betweenAsterisk1.4beta2 and Aastra 480i
> 
> 
> On 10/6/06, Gareth Owen <gowen at aastra.com> wrote:
> Morten,
> 
> Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to
> see what is going on.  Can you post the INVITE message that is being
> rejected?
> 
> 
> This INVITE results in a 488 from the phone:
> 
> INVITE sip:1014 at 192.168.10.100 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK42f78e77;rport
> From: "1011" < sip:1011 at 192.168.10.2>;tag=as3a35aa3a
> To: <sip:1014 at 192.168.10.100>
> Contact: < sip:1011 at 192.168.10.2>
> Call-ID: 15467e4462b5620e1e7155e96a5dc0ba at 192.168.10.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 06 Oct 2006 14:22:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 309
> v=0
> o=root 4746 4746 IN IP4 192.168.10.2
> s=session
> c=IN IP4 192.168.10.2
> t=0 0
> m=audio 10066 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:3 GSM/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
> And this INVITE works (only alaw is enabled):
> INVITE sip:1014 at 192.168.10.100 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK3c04692a;rport
> From: "1011" < sip:1011 at 192.168.10.2>;tag=as39cd0724
> To: <sip:1014 at 192.168.10.100>
> Contact: < sip:1011 at 192.168.10.2>
> Call-ID: 32a8f09a785b36cf5e8b6ba02b5afb00 at 192.168.10.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 06 Oct 2006 14:23:51 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 238
> v=0
> o=root 4762 4762 IN IP4 192.168.10.2
> s=session
> c=IN IP4 192.168.10.2
> t=0 0
> m=audio 10042 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
> 
> Also, I know we've fixed a number of SDP related issues in 1.4.1, so if
> you haven't already you might want to try the 1.4.1 beta.  Info on how to
> get the beta is available here:
> 
> http://groups.google.com/group/Aastra-480i-
> Users/browse_frm/thread/8f6f0f3419ef396d
> 
> 
> I will try that and report back here.
> 
> 
> --
> Morten Isaksen
> http://www.misak.dk/blog/
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users