asterisk users - Sep 2013 - Is this SDP payload Asterisk created valid?

We have an issue with a customer where when calls are sent to one of their
offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the server which
initiated the call to our switch which is on the same network.
After the call is answered we switch the audio which is accepted fine but we
then send the following packet and get a SIP/488 response from the far end.
This packet seems to be updating the version for the o= session id which is fair
enough. Its sending the c= connection data but not the m=audio line
which appears to be what the remote end is complaining about.

Can anyone with a bit more knowledge about the SDP standard tell me if what
asterisk is doing is correct?
Or if it must be a bug with our customers equipment?

Thanks
Gareth

U 2013/09/27 11:04:55.352854 88.x.x.25:5060 ->  213.x.x.24:5060
INVITE sip:0844xxxxxx at 146.x.x.10:54900 SIP/2.0.
Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713.
Route:<sip:213.x.x.24;lr=on;ftag=as691af817;did=ecd.c2dc96e6>.
Max-Forwards: 70.
From:<sip:01628xxxxxx at 88.x.x.25>;tag=as691af817.
To:<sip:0844xxxxxx at freespeech.co.uk>;tag=ee7a6c7cad57f096i1.
Contact:<sip:01628xxxxxx at 88.x.x.25:5060>.
Call-ID: 2eeb643d086234de59a1fd4e78170d3f at 88.x.x.25:5060.
CSeq: 104 INVITE.
User-Agent: Asterisk PBX 11.2-cert2.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 110.
.
v=0.
o=root 716216031 716216033 IN IP4 88.x.x.35.
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.35.
t=0 0.

#
U 2013/09/27 11:04:55.365458 213.x.x.24:5060 ->  88.x.x.25:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713;rport=5060.
From:<sip:01628xxxxxx at 88.x.x.25>;tag=as691af817.
To:<sip:0844xxxxxx at freespeech.co.uk>;tag=ee7a6c7cad57f096i1.
Call-ID: 2eeb643d086234de59a1fd4e78170d3f at 88.x.x.25:5060.
CSeq: 104 INVITE.
Server: OpenSIPS (1.5.3-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 2013/09/27 11:04:55.431674 213.x.x.24:5060 ->  88.x.x.25:5060
SIP/2.0 488 Not Acceptable Here.
To:<sip:0844xxxxxx at freespeech.co.uk>;tag=ee7a6c7cad57f096i1.
From:<sip:01628xxxxxx at 88.x.x.25>;tag=as691af817.
Call-ID: 2eeb643d086234de59a1fd4e78170d3f at 88.x.x.25:5060.
CSeq: 104 INVITE.
Via: SIP/2.0/UDP
88.x.x.25:5060;rport=5060;received=88.x.x.25;branch=z9hG4bK62215713.
Record-Route:<sip:213.x.x.24;lr=on;ftag=as691af817>.
Contact: "freespeech"<sip:0844xxxxxx at 146.x.x.10:54900>.
Warning: 304 spa "Media type not available".
Server: Cisco/SPA303-7.5.4.
Content-Length: 0.

On 27/09/13 14:15, Gareth Blades wrote:
>
> Can anyone with a bit more knowledge about the SDP standard tell me if 
> what asterisk is doing is correct?
> Or if it must be a bug with our customers equipment? 
Reading RFC2327 it cais the c= line 'must' be present in all updates 
while 'm=' media lines are optional.
I am therefore inclined to believe that Asterisk is working correctly 
and there is a bug in the customers SIP equipment.
But thats just my personal interpretation from briefly reading the standard.

Gareth Blades wrote:
> We have an issue with a customer where when calls are sent to one of
> their offices as soon as the call is answered the call fails.
> We are performing remote bridging and switching the audio from the
> server which initiated the call to our switch which is on the same network.
> After the call is answered we switch the audio which is accepted fine
> but we then send the following packet and get a SIP/488 response from
> the far end.
> This packet seems to be updating the version for the o= session id which
> is fair enough. Its sending the c= connection data but not the m=audio line
> which appears to be what the remote end is complaining about.
>
> Can anyone with a bit more knowledge about the SDP standard tell me if
> what asterisk is doing is correct?
> Or if it must be a bug with our customers equipment?
The SDP you posted should be fine BUT my question becomes... have you 
modified chan_sip at all? I don't think it should be possible for it to 
not put any media lines in...

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org