7 Feb
2015
7 Feb
'15
2:24 p.m.
I think i have similar problem last week with rtpengine deployment which
was about 1-2 weeks old. There was no audio although the logs say that STUN
bindings are successful from both side (SAVPF <-> AVP). One symptom of the
problem is this log message,
--
rtpengine[16455]: [kr8shv3uca0fnmg4ktd4 port 41198] SRTP output wanted, but
no crypto suite was negotiated
--
As a last resort before filing an official bug, i decided to upgrade
RTPEngine which seems to have solved the problem. Interestingly both
rtpengine deployments (the one causing no audio and the upgraded one) have
the same version numbers (i.e. the output of "rtpengine -v"), so i can't
actually pinpoint which revision has this problem and which one has solved
this problem. Anyhow, can you also try with latest RTPEngine from official
git repo and see if that solves the problem. The git commit number of
working RTPEngine is 9a2da87f130ab3c1e21d9b593efec78a8eb7b3f3 (ah, i miss
subversion which has more meaningful linear revision numbers ...).
For RTPEngine developers, can you guys add git revision string as extended
version for "rtpengine -v" output? It may be the same way as kamailio does,
e.g.
kamailio 4.2.2 (i386/linux) *6f7306*
This will tremendously help in tracking bugs and their fixes.
Thank you.
On Fri, Feb 6, 2015 at 5:07 PM, Don Fanning <don(a)00100100.net> wrote:
I guess I should also note that I do have freeswitch
on the machine and
the contexts are there. It seems to work fine in echotest and conference
modes. But I do have to caveat that I have not tried it recently with some
recent config changes to handle the RTP/SAVPF to RTP/AVP rewriting - so
that I cannot be sure of.
On Fri, Feb 6, 2015 at 5:47 AM, Fred Posner <fred(a)palner.com> wrote:
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
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Hello All,
I currently am running Kamailio in a WSS configuration with sipML5. I
use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the
destination server only supports AVP or SAVP.
RTPEngine has no issues rewriting the packet going out, the SIP session
comes up and handshakes correctly to start the session. Then the remote
server sends the RTP stream back and I'm having issues getting Kamailio
or RTPEngine or something to take the RTP stream, process it back to
SAVPF and send it back out the WSS port. I see no RTP data come from
Kam/RTPengine going towards the remote server.
On the server side, I do see STUN being called and the sipML5 bind
successfully but that's it.
Both the destination and server sit without firewalls in the way so the
problem has to be what I'm doing in Kamailio.
My config is located here:
http://pastebin.com/dWLdUz5j
Packet dumps available upon request.
Thanks for any assistance!
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