Reported by Taras S <vinni.jopa@gmail.com> on Aug 5, 2010
see
http://groups.google.com/group/myprojectguide/browse_thread/thread/4449bb7c9b48f
bae/cb25671d2e0acd39?pli=1
hi
I used siprtmp and I have a question about it
If I'm calling(using speex8) from flash to asterisk, on music on hold
extensions, sound quality is awful.(in trafshow seen that rtp flow is
unstable. From time to time the flow stops and then resumes again,
this leads to failures in the sound). But if i calling(using speex8)
from flash to freeswitch, on music on hold extensions, sound quality
are normal(steady stream of rtp). I'm confused.
--
There were some timestamp related bug fixes done in version 6.0 of
rtmplite/siprtmp. Can you please give it a try.
If you are already trying version 6.0 or later, could you please
enable RTP trace (set _debug=True in siprtmp.py, rtmplite.py and in
p2p-sip's rfc3550.py) and send the trace as reply. Basically I want to
see the RTP timestamp and actual RTMP time when packets are sent by
asterisk to Flash application.
Thanks.
---
hi
This is tarce log siprtmp ver6.0
Original issue reported on code.google.com by voiprese...@gmail.com on 5 Feb 2011 at 2:18
Original issue reported on code.google.com by
voiprese...@gmail.comon 5 Feb 2011 at 2:18