When running wireshark with filter udp.dstport = 5060 i dont see the BYE frame. I only see: source dest prot info 10.0.1.19 10.0.1.30 SIP/SPD Request: INVITE sip:3200 at 10.0.1.30<sip%3A3200 at 10.0.1.30>, with session description 10.0.1.30 10.0.1.19 SIP Status: 180 Ringing 10.0.1.30 10.0.1.19 SIP Status: 200 OK, with session description 10.0.1.19 10.0.1.30 SIP Request: ACK sip:10.0.1.30:5060 I dont see anymore frames regarding SIP session between those 2 hosts. However, checking the logs on the destination 3rd party product (10.0.1.30) Received ACK Sent Command: BYE Received Message: STATUS Received STATUS msg after BYE(200) sendMessageToRtpProvider <CHANNEL_HANGUPPED_> After this happens on the destination endpoint, on pjsip i start to receive spam of: RTP recv() error: Connection reset by peer (WSAECONNRESET) On Tue, Jun 3, 2008 at 11:22 AM, Benny Prijono <bennylp at pjsip.org> wrote: > On Tue, Jun 3, 2008 at 10:51 AM, Jo?o C?sar <jpcesar at gmail.com> wrote: > > Hell all > > > > I'm working on a project that uses pjsip, pjmedia and pjsua-api. 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 need to implement explicit hangup here if connection is no longer active? > > > > No you don't need to do anything in on_call_state(), as call > disconnection will be handled automatically and on_call_state() only > serves to report the changes in call state. In your case above, I > suspect that the pjsip doesn't receive the BYE request from the other > side. Please check the log and/or Wireshark packet capture to see if > this is/isn't the case. > > Cheers > Benny > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- 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/20080603/991c090c/attachment-0001.html
