good day, i have problem in RTC Client 1.3 HOLD control when using softphone(rtc) with softphone(rtc), i cannot hold the call but when using softphone(rtc) with sipphone/ip phone , i can hold the call but once unhold the call, both side cannot hear the voice i check the sip message, i found that the port number has been changed, why port number will change? is there any solution so that both side can hear voice after unhold? i test samples provided by ms, they can hold the call, but those application is direct ip to ip call, and the sip message also shown that the port number has been changed. does anyone involve in RTC Client development before and face this problem also? call 01548408907: INVITE sip:01548408907@218.189.19.37 SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 1 INVITE Contact: <sip:192.228.118.173:11564> User-Agent: RTC/1.3 Content-Type: application/sdp Content-Length: 291 v=0 o=- 0 0 IN IP4 192.228.118.173 s=session c=IN IP4 192.228.118.173 b=CT:1000 t=0 0 m=audio 16384 RTP/AVP 97 0 8 4 101 a=rtpmap:97 red/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:rejected trying 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 1 INVITE Server: Sip EXpress router (0.9.0 (i386/linux)) Content-Length: 0 Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19331 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:01548408907@218.189.19.37 out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1" ringing 180 Ringing Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on> Contact: <sip:01548408907@192.228.118.223:5060> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 1 INVITE Content-Length: 0 accept call 200 OK Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on> Contact: <sip:01548408907@219.94.42.174:5060> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 265 v=0 o=01548408907 514117283 514117283 IN IP4 192.228.118.223 s=CrystalMedia Session c=IN IP4 218.189.19.37 t=0 0 m=audio 36456 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:60001 a=direction:both a=sendrecv ack from me ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 1 ACK Route: <sip:01548408907@219.94.42.174:5060> User-Agent: RTC/1.3 Content-Length: 0 i hold the call now ?INVITE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 2 INVITE Route: <sip:01548408907@219.94.42.174:5060> Contact: <sip:192.228.118.173:11564> User-Agent: RTC/1.3 Content-Type: application/sdp Content-Length: 168 v=0 o=- 0 0 IN IP4 0.0.0.0 s=session c=IN IP4 0.0.0.0 t=0 0 m=audio 16384 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 trying 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 2 INVITE Server: Sip EXpress router (0.9.0 (i386/linux)) Content-Length: 0 Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19326 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1" success hold call 200 OK Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on> Contact: <sip:01548408907@219.94.42.174:5060> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 2 INVITE Content-Type: application/sdp Content-Length: 173 v=0 o=01548408907 1442463500 1442463500 IN IP4 192.228.118.223 s=CrystalMedia Session c=IN IP4 0.0.0.0 t=0 0 m=audio 36456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly ack ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 2 ACK Route: <sip:01548408907@219.94.42.174:5060> User-Agent: RTC/1.3 Content-Length: 0 unhold call now INVITE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 3 INVITE Route: <sip:01548408907@219.94.42.174:5060> Contact: <sip:192.228.118.173:11564> User-Agent: RTC/1.3 Content-Type: application/sdp Content-Length: 195 v=0 o=- 0 0 IN IP4 192.228.118.173 s=session c=IN IP4 192.228.118.173 b=CT:1000 t=0 0 m=audio 34046 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 trying 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 3 INVITE Server: Sip EXpress router (0.9.0 (i386/linux)) Content-Length: 0 Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19326 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1" success unhold call 200 OK Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on> Contact: <sip:01548408907@219.94.42.174:5060> Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 3 INVITE Content-Type: application/sdp Content-Length: 209 v=0 o=01548408907 541153517 541153517 IN IP4 192.228.118.223 s=CrystalMedia Session c=IN IP4 218.189.19.37 t=0 0 m=audio 36456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=rtcp:60001 a=direction:both a=sendrecv ack from me ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 3 ACK Route: <sip:01548408907@219.94.42.174:5060> User-Agent: RTC/1.3 Content-Length: 0 bye message BYE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.228.118.173:11564 Max-Forwards: 70 From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84 To: <sip:01548408907@218.189.19.37>;tag=6489fa9e Call-ID: 938d582d1cec4493a7a4dadcf688945d CSeq: 4 BYE Route: <sip:01548408907@219.94.42.174:5060> User-Agent: RTC/1.3 Content-Length: 0 ............... these are the sip message can someone please figure out for me why both side cannot hear the voice after unhold call second question is about CoCreateInstance if i have 2 dll files both also use CoInitializeEx will the 1st dll file affect the 2nd one? because i always CoCreateInstance fails i try on 2 application, one is onli 1 dll, another one is more than one, some of them may b using CoInitializeEx function the 1st application can work but the 2nd one cannot work does anybody know this problem and teach me solve this problem? thanks a lot the attachment file is txt file but the original file is saved from ethereal rtc_log.txt
