Hi, I?m having a connectivity problem with pjsip beta 2. I have SIP and TURN servers setup using reSiprocate. When the ICE negotiation process selects the server reflexive candidates, we get audio/video between the two clients for 30 seconds or so and then the SIP server sends out a ?flow failed? message and the call is terminated. However, using the same client machines, the ICE negotiation process occasionally selects the TURN relay candidates. When this happens, the audio/video work perfectly and the call stays active until manually hung-up. While I try to track down why calls consistently fail with the server reflexive candidates, is there a way to force pjsip to use the TURN relay? I checked the pjsua and pjnath headers and didn?t see any obvious way to force use of the TURN relay... Regards, --Jonathan Martin -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120224/c9669213/attachment.html>
