[asterisk-users] Is this SDP payload Asterisk created valid?

[Joshua Colp]

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