Hi Benny, I had the following problem: My PJSUA-like application is based on release 0.8.0. After it has cleared a call, the opposite involved SIP-phone kept on sending RTP (due to some error condition elsewhere). After answering the next call my application switches the RTP address alternatingly between the old and new party: udpmedia Remote RTP address switched to 192.168.9.83:49226 udpmedia Remote RTP address switched to 192.168.9.72:49204 ... I found in the mail-archive that this can be solved in my case by: supplying option PJMEDIA_UDP_NO_SRC_ADDR_CHECKING for pjmedia_transport_udp_attach in pjsua_media.c This works for my application, although the old RTP stream is troubling causing "jitter buffer reset". Now my question is: Can the option PJMEDIA_UDP_NO_SRC_ADDR_CHECKING be made available via config_site.h ? This way it is easier to include new developments to pjproject in my application. Best regards, Arie Velthoen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080227/e65fa363/attachment-0001.html
