[asterisk-users] Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i

[Gareth Owen]

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/