Exception in siprtmp when remote side removed m=video in answer SDP

First reported by Tom van der Geer <tom.vandergeer@xs4all.nl> on Jun 24, 2011.

Hi,

I started using rtmplite today and managed to get it working. I'm
particularly interested in using rtmplite together with a VoIP gateway
to call to a regular PSTN line. I've managed to get speex transcoding
working as well.
However... When the call is established a lot of errors are scrolling
in my window:

exception in rtmp_data (<type 'exceptions.TypeError'>,
TypeError("'NoneType' object is unsubscriptable",), <traceback object
at 0x89e7914>)

These seem to be related to audio being sent from the Flash client to
the siprtmp-gw, because when it detects 'silence' from my mic the
scrolling of the error messages stops...

There's no audio from the phone played on the Flash client side at
all.

One more thing to point out. When I press the "H" button on the Flash
client (I suppose that means "Hold") it does play an audio file on my
phone. So that works!

So, it looks like the signalling and transcoding work, but the media
between the siprtmp-gw and the flash client is not working.

Any hints/tips/suggestions?

TIA,

Tom

--

I fixed it in p2p-sip's svn r38.
http://code.google.com/p/p2p-sip/source/detail?r=38

The problem was that your receiving SIP user agent is sending
incorrect SDP in the answer. As per offer/answer model of SIP in RFC
3264, the answer SDP should have all the m= lines that were in offer,
but with port of 0 if the receiver doesn't support that media. In the
response from your SIP user agent, the SDP has missing m= line for
video.

I fixed p2p-sip's voip.py module to work around this case.

Hope this helps!

Original issue reported on code.google.com by kundan10 on 26 Jun 2011 at 8:04

ashumeow / p2p-sip

Exception in siprtmp when remote side removed m=video in answer SDP #3