Hi, I have set transport = tls option in sip uri. Though transport till 200 OK is on TLS but ACK sent from called side on tcp and following observations are present : 1. Invite received on other side is on TLS port 5061 have Contact = <sip:111.111.187.175:5061;transport=tcp and not <sip:111.111.187.175:5061;transport=tls 2. ACK sent after 200 OK is on tcp, so it couldnot reach on the called side which is expecting tls transport due to which calling side end up sending 200 OK 8, 9 times before dropping the call by sending BYE. ACK in response of 200 OK must also be SENT via tls transport. CALLING SIDE logs 11:19:22.883 ? ?pjsua_acc.c ?....sip:12342061 at 111.111.187.175: registration success, status=200 (OK), will re-register in 60 seconds 11:19:49.225 ? pjsua_core.c ?.RX 1092 bytes Request msg INVITE/cseq=647024895 (rdata0x1503d38) from TLS 111.111.187.175:5061: INVITE sip:12342061 at 111.111.1.171:57777;transport=TLS;ob SIP/2.0 Via: SIP/2.0/TLS 111.111.187.175:5061;branch=z9hG4bK+6eb56f78ddba93b284d6648da5b643601+SBC+1 From: "12342065" <sip:12342065 at 111.111.187.175:5061>;tag=111.111.187.175+1+7724a7c7+c8d96d02 To: <sip:12342061 at 111.111.1.171:57777;transport=TLS;ob> CSeq: 647024895 INVITE Expires: 180 Organization: c Content-Length: 476 Contact: <sip:111.111.187.175:5061;transport=tcp> Content-Type: application/sdp Max-Forwards: 69 Call-ID: 5cf4005bc99e4c0e9ccf47f9ac0f6c52 at 111.111.187.175 Accept: application/sdp, application/dtmf-relay v=0 o=- 3659233868 3659233868 IN IP4 111.111.1.186 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 111.111.1.186 b=TIAS:64000 a=rtcp:4003 IN IP4 111.111.1.186 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 11:19:49.225 ? pjsua_call.c ?.Incoming Request msg INVITE/cseq=647024895 (rdata0x1503d38) 11:19:49.225 ?pjsua_media.c ?..Call 0: initializing media.. 11:19:49.225 ?pjsua_media.c ?...RTP socket reachable at 111.111.1.171:4000 11:19:49.225 ?pjsua_media.c ?...RTCP socket reachable at 111.111.1.171:4001 11:19:49.225 ?pjsua_media.c ?...Media index 0 selected for audio call 0 11:19:49.226 ? pjsua_core.c ?.....TX 382 bytes Response msg 100/INVITE/cseq=647024895 (tdta0x7f93e0045e60) to TLS 111.111.187.175:5061: SIP/2.0 100 Trying Via: SIP/2.0/TLS 111.111.187.175:5061;received=111.111.187.175;branch=z9hG4bK+6eb56f78ddba93b284d6648da5b643601+SBC+1 Call-ID: 5cf4005bc99e4c0e9ccf47f9ac0f6c52 at 111.111.187.175 From: "12342065" <sip:12342065 at 111.111.187.175>;tag=111.111.187.175+1+7724a7c7+c8d96d02 To: <sip:12342061 at 111.111.1.171;ob> CSeq: 647024895 INVITE Content-Length: ?0 --end msg-- 11:19:49.226 ? ?pjsua_aud.c ?..Conf connect: 2 --> 0 11:19:49.226 ? ?pjsua_aud.c ?...Set sound device: capture=-1, playback=-2 11:19:49.226 ? ?pjsua_app.c ?....Turning sound device ON 11:19:49.226 ? ?pjsua_aud.c ?....Opening sound device PCM at 16000/1/20ms 11:19:49.311 ? ?pjsua_app.c ?....Turning sound device ON 11:19:49.311 ? ?pjsua_aud.c ?....Opening sound device PCM at 44100/1/20ms 11:19:49.380 ec0x7f93e004bf ?.....AEC created, clock_rate=44100, channel=1, samples per frame=882, tail length=200 ms, latency=0 ms 11:19:49.380 ? conference.c ?...Port 2 (ring) transmitting to port 0 (HDA Intel MID: 92HD75B3X5 Analog (hw:0,0) (44KHz)) 11:19:49.380 ? ?pjsua_app.c ?..Incoming call for account 0! Media count: 1 audio & 0 video From: "12342065" <sip:12342065@111.111.187.175> To: <sip:12342061 at 111.111.1.171;ob> Press a to answer or h to reject call 11:19:49.403 os_core_unix.c ?Info: possibly re-registering existing thread Answer with code (100-699) (empty to cancel): 11:19:51.923 pjsua_call.c !Answering call 0: code=200 11:19:51.923 ?pjsua_media.c ?...Call 0: updating media.. 11:19:51.923 ? ?pjsua_aud.c ?....Audio channel update.. 11:19:51.923 ?strm0x15079d8 ?.....VAD temporarily disabled 11:19:51.923 ?strm0x15079d8 ?.....Encoder stream started 11:19:51.923 ?strm0x15079d8 ?.....Decoder stream started 11:19:51.923 ?pjsua_media.c ?....Audio updated, stream #0: speex (sendrecv) 11:19:51.923 ? ?pjsua_app.c ?...Call 0 media 0 [type=audio], status is Active 11:19:51.923 ? ?pjsua_aud.c ?...Conf disconnect: 2 -x- 0 11:19:51.923 ? conference.c ?....Port 2 (ring) stop transmitting to port 0 (HDA Intel MID: 92HD75B3X5 Analog (hw:0,0) (44KHz)) 11:19:51.923 ? ?pjsua_aud.c ?...Conf connect: 3 --> 0 11:19:51.923 ? conference.c ?....Port 3 (sip:12342065 at 111.111.187.175:5061) transmitting to port 0 (HDA Intel MID: 92HD75B3X5 Analog (hw:0,0) (44KHz)) 11:19:51.923 ? ?pjsua_aud.c ?...Conf connect: 0 --> 3 11:19:51.923 ? conference.c ?....Port 0 (HDA Intel MID: 92HD75B3X5 Analog (hw:0,0) (44KHz)) transmitting to port 3 (sip:12342065 at 111.111.187.175:5061) 11:19:51.923 ? pjsua_core.c ?....TX 926 bytes Response msg 200/INVITE/cseq=647024895 (tdta0x7f93e0045e60) to TLS 111.111.187.175:5061: SIP/2.0 200 OK Via: SIP/2.0/TLS 111.111.187.175:5061;received=111.111.187.175;branch=z9hG4bK+6eb56f78ddba93b284d6648da5b643601+SBC+1 Call-ID: 5cf4005bc99e4c0e9ccf47f9ac0f6c52 at 111.111.187.175 From: "12342065" <sip:12342065 at 111.111.187.175>;tag=111.111.187.175+1+7724a7c7+c8d96d02 To: <sip:12342061 at 111.111.1.171;ob>;tag=2c76fc04-2ca8-400f-90e9-d7e13e956c01 CSeq: 647024895 INVITE Contact: <sip:111.111.1.171:5060;transport=TCP> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: ? 278 v=0 o=- 3659233789 3659233790 IN IP4 111.111.1.171 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 96 c=IN IP4 111.111.1.171 b=TIAS:64000 a=rtcp:4001 IN IP4 111.111.1.171 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 11:19:51.924 ? ?pjsua_app.c ?.......Call 0 state changed to CONNECTING >>> 11:19:52.424 ? pjsua_core.c ?.TX 926 bytes Response msg 200/INVITE/cseq=647024895 (tdta0x7f93e0045e60) to TLS 111.111.187.175:5061: Please help me regards shubham
