Sorry for missing this in my last post, but I’m using sofia for all calls.

/Peter

Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Peter Olsson
Skickat: den 29 maj 2009 12:31
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn’t upgraded until today, so I don’t know exactly when this happened.

I’ve noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it’s somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server.

The conf-dir is the same between the revisions.

The calls I’m trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS).

Regards,

Peter

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