[SR-Users] One Way RTP from destination SIP server using rtpengine

[Richard Fuchs]

On 07/02/15 07:24 AM, Muhammad Shahzad wrote:
> 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.

It does, but it only works if the build environment has git available.
Otherwise it has to be left out.

tabasco:~/src/ngcp-git/rtpengine/daemon(master)# ./rtpengine -v
3.3.0.0+0~mr3.8.0.0 git-master-9a2da87

cheers