Hello. First of all want to congratulate all those who worked on this stack, it's an awesome work so far. I'm working on a project that uses pjsip, pjmedia and pjsua. We have a memory port as source of audio, and we are establishing a sip connection to a SIP compliant third party product. However, on call termination from the other side, i think we are not properly handing the hangup of the call since we are getting this kind of errors: strm0284c64c RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] strm0284c64c RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] strm0284c64c RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] assumptions: - we are using UDP transport - we are adding the ports with pjsua_conf_add_port - making the call with pjsua_call_make_call - using pjmedia_snd_port_create_player to create the port - using pjmedia_snd_port_connect to connect the ports we have implemented the following callback functions - on_call_media_state - on_call_state - on_incoming_call However, on the on_call_state we are only printing some call info. do we need to implement explicit hangup here if connection is no longer active? Best regards -- Joao Cesar msn: jpcesar at gmail.com gtalk: jpcesar at gmail.com icq: 13790802 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080603/ec0669ed/attachment.html
