Hi all, I came across a problem, that I'd like some help with, as I don't seem to be able to find the solution. I'm on pjsip 1.14.2-svn (latest svn for ver 1), and the problem I have is this: I have a nic in my client with IP 192.168.0.100, and set up a vpn connection to workplace where I get a dhcp assigned IP in the 192.168.4.x subnet. Now, if I don't specify the public_addr in the SIP/RTP transport_config, SIP/RTP socket is initially reachable at 192.168.0.100: pjsua_core.c SIP UDP socket reachable at 192.168.0.100:5030 udp07C86CB8 SIP UDP transport started, published address is 192.168.0.100:5030 pjsua_media.c RTP socket reachable at 192.168.0.100:8770 pjsua_media.c RTCP socket reachable at 192.168.0.100:8771 pjsua_media.c RTP socket reachable at 192.168.0.100:8772 pjsua_media.c RTCP socket reachable at 192.168.0.100:8773 And after I send the registration, SIP transport correctly adjusts itself to the VPN address: pjsua_acc.c IP address change detected for account 0 (192.168.0.100:5030 --> 192.168.4.110:5030). Updating registration (using method 2) ... while RTP socket is still at 192.168.0.100 which can be seen in SDP SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.199:5060;received=192.168.4.199;branch=z9hG4bK639b71a6 Call-ID: 4660c5d37f2fbe50309f550a2fc07387 at 192.168.4.199:5060 From: "Unknown" <sip:Unknown@192.168.4.199>;tag=as52a4bb8b To: <sip:700 at 192.168.4.110;ob>;tag=z9hG4bK639b71a6 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer Content-Type: application/sdp Content-Length: 442 v=0 o=- 3559047092 3559047092 IN IP4 192.168.0.100 s=pjmedia c=IN IP4 192.168.0.100 t=0 0 m=audio 8766 RTP/AVP 0 8 98 97 3 99 104 9 96 a=rtcp:8767 IN IP4 192.168.0.100 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ... which basically means, that while I can initiate and hangup the call, audio never gets through. Now - I'm looking for a way to catch the address change, so I can update the RTP transport address. I guess the best way would be to do that in 'on_reg_state', once the registration is successful, as at that time SIP transport already is updated to the IP of my correct NIC (vpn in this case), but somehow I couldn't find a way to get the new SIP transport address. Any suggestions? Greets, Toni ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.
