Hi all, I'm using PJSIP with TCP transport and a SIP trunk(registration over TCP, SIP calls over TCP) to make calls to a mobile phone. The environment is something like this PJSIP <------ SIP over TCP ------> SIP Server <--- SIP trunk --> SIP trunk Provider <---PSTN/Mobile Gateway-->Mobile phone All calls are made from PJSIP (pjsua_app.c) over TCP to the mobile. To disable UDP transport creation I inserted a line "cfg->no_udp = true;" at the end of the function? "static void default_config(struct app_config *cfg)" I followed the instructions given in http://trac.pjsip.org/repos/wiki/Using_SIP_TCP to make calls over TCP. The problem is that we don't receive RTP media from the mobile end into PJSIP.? But RTP packets from the PJSIP dialer reach mobile side just fine. We can hear audio in the mobile end. We found from packet traces that the reason we dont receive media is that the SIP server is sending RTP packets to a private IP address. But when we switch to UDP for registration and send INVITES over TCP the call works fine (audio at both ends). This is what the wireshark capture shows when we use TCP only for all SIP messages - 1. PJSIP registers with server over TCP. 2. Server sends 401 with PJ's public IP and port in VIA 3. PJ registers again but inserts its public ip and port in the contact header in the next REGISTER message sequence. So far so good. Same sequence of messages seen when UDP is used to REGISTER. 4. INVITE sent over TCP. Dialog establishment works fine. But in the record-route header nat=yes is missing. Server sends media to private IP. No media received at PJSIP. Is this a bug in PJSIP? If so how can this be fixed. Wireshark packet traces are available on request. Your help and inputs are much appreciated. Best Regards, John -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130822/86cabb14/attachment-0001.html>
