I also attach source code (not difference from http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm) When start program with "myapp.exe sip:2222 at 192.168.1.106<sip%3A2222 at 192.168.1.106>" it work well this program make call to another phone. hangup from 2222 is ok But make call from 2222 to this program then hang it up , it crash (program terminate) I don't know what wrong because i copy this code from http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm I compile myapp.cpp with MinGW under Windows 7. Anyone can help? Thank. Last message before crash --end msg-- 11:38:06.769 APP Call 1 state=CONFIRMED 11:38:06.809 ec0x1a9d688 Underflow, buf_cnt=0, will generate 1 frame 11:38:07.289 strm0x1a991b4 VAD re-enabled 11:38:28.819 pjsua_core.c RX 535 bytes Request msg OPTIONS/cseq=102 (rdata0x1a88864) from UDP 192.168.1.106:5060: OPTIONS sip:1111 at 192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK2b9f16c4;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@192.168.1.106 <sip%3AUnknown at 192.168.1.106> >;tag=as577732a7 To: <sip:1111 at 192.168.1.100:5060> Contact: <sip:Unknown at 192.168.1.106 <sip%3AUnknown at 192.168.1.106>> Call-ID: 26ae8aee33ec93a37ca4095a663299d0 at 192.168.1.106 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Date: Wed, 19 Aug 2009 04:38:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --end msg-- 11:38:34.053 Master/sound Underflow, buf_cnt=0, will generate 1 frame -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090819/bec057e4/attachment-0001.html> -------------- next part -------------- /* #include <pjlib.h> #include <pjlib-util.h> #include <pjmedia.h> #include <pjmedia-codec.h> #include <pjsip.h> #include <pjsip_simple.h> #include <pjsip_ua.h> #include <pjsua-lib/pjsua.h> */ #include <pjsua-lib/pjsua.h> #define THIS_FILE "APP" #define SIP_DOMAIN "192.168.1.106" #define SIP_USER "1111" #define SIP_PASSWD "1111" static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata){ pjsua_call_info ci; PJ_UNUSED_ARG(acc_id); PJ_UNUSED_ARG(rdata); pjsua_call_get_info(call_id, &ci); PJ_LOG(3,(THIS_FILE, "Incomming call from %.*s!!", (int)ci.remote_info.slen, ci.remote_info.ptr)); pjsua_call_answer(call_id, 200, NULL, NULL); } static void on_call_state(pjsua_call_id call_id, pjsip_event *e){ pjsua_call_info ci; PJ_UNUSED_ARG(e); pjsua_call_get_info(call_id, &ci); PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id, (int) ci.state_text.slen, ci.state_text.ptr)); } static void on_call_media_state(pjsua_call_id call_id){ pjsua_call_info ci; pjsua_call_get_info(call_id, &ci); if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { pjsua_conf_connect(ci.conf_slot, 0); pjsua_conf_connect(0, ci.conf_slot); } } static void error_exit(const char *title, pj_status_t status){ pjsua_perror(THIS_FILE, title, status); pjsua_destroy(); exit(1); } int main(int argc, char *argv[]) { pjsua_acc_id acc_id; pj_status_t status; status = pjsua_create(); if(status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status); if(argc >1){ status = pjsua_verify_sip_url(argv[1]); if(status != PJ_SUCCESS) error_exit("Invalid URL in argv", status); } { pjsua_config cfg; pjsua_logging_config log_cfg; pjsua_config_default(&cfg); cfg.cb.on_incoming_call = &on_incoming_call; cfg.cb.on_call_media_state = &on_call_media_state; cfg.cb.on_call_state = &on_call_state; pjsua_logging_config_default(&log_cfg); log_cfg.console_level = 4; status = pjsua_init(&cfg, &log_cfg, NULL); if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status); } /* Add UDP transport.*/ { pjsua_transport_config cfg; pjsua_transport_config_default(&cfg); cfg.port = 5060; status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL); if(status != PJ_SUCCESS) error_exit("Error creating transport", status); } /* Initialization is done, now start pjsua */ status = pjsua_start() ; if(status != PJ_SUCCESS) error_exit("Error start pjsua", status); /* Register to SIP server by creating SIP account. */ { pjsua_acc_config cfg; pjsua_acc_config_default(&cfg); cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN); cfg.reg_uri = pj_str("sip:" SIP_DOMAIN); cfg.cred_count = 1 ; cfg.cred_info[0].realm = pj_str("*"); cfg.cred_info[0].scheme = pj_str("digest"); cfg.cred_info[0].username = pj_str(SIP_USER); cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD; cfg.cred_info[0].data = pj_str(SIP_PASSWD); status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id); if(status != PJ_SUCCESS) error_exit("Error adding account", status); } /* If URI is specified , make call to the URL. */ if(argc > 1){ pj_str_t uri = pj_str(argv[1]); status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL); if(status != PJ_SUCCESS) error_exit("Error making call", status); } /* Wait util user press "q" to quit. */ for(;;){ char option[10]; puts("Press 'h' to hangup all calls, 'q' to quit"); if(fgets(option, sizeof(option), stdin) == NULL){ puts("EOF while reading stdin, will quit now.."); break; } if(option[0] == 'q') break; if(option[0] == 'h') pjsua_call_hangup_all(); } pjsua_destroy(); return 0; } -------------- next part -------------- 11:42:27.851 os_core_win32. pjlib 1.3 for win32 initialized 11:42:27.875 sip_endpoint.c Creating endpoint instance... 11:42:27.875 pjlib select() I/O Queue created (0x39bca0) 11:42:27.875 sip_endpoint.c Module "mod-msg-print" registered 11:42:27.875 sip_transport. Transport manager created. 11:42:27.875 sip_endpoint.c Module "mod-pjsua-log" registered 11:42:27.875 sip_endpoint.c Module "mod-tsx-layer" registered 11:42:27.875 sip_endpoint.c Module "mod-stateful-util" registered 11:42:27.875 sip_endpoint.c Module "mod-ua" registered 11:42:27.875 sip_endpoint.c Module "mod-100rel" registered 11:42:27.876 sip_endpoint.c Module "mod-pjsua" registered 11:42:27.876 sip_endpoint.c Module "mod-invite" registered 11:42:27.917 pa_dev.c PortAudio sound library initialized, status=0 11:42:27.917 pa_dev.c PortAudio host api count=2 11:42:27.917 pa_dev.c Sound device count=5 11:42:27.923 wmme_dev.c WMME initialized, found 4 devices: 11:42:27.923 wmme_dev.c dev_id 0: Wave mapper (in=2, out=2) 11:42:27.923 wmme_dev.c dev_id 1: Microphone (High Definition Aud (in=2, out=0) 11:42:27.923 wmme_dev.c dev_id 2: Line In (High Definition Audio (in=2, out=0) 11:42:27.923 wmme_dev.c dev_id 3: Speakers (High Definition Audio (in=0, out=2) 11:42:27.924 pjlib select() I/O Queue created (0x1b59d6c) 11:42:27.924 libsrtp Ugh: /dev/urandom not present, using rand() instead 11:42:27.926 libsrtp Ugh: /dev/urandom not present, using rand() instead 11:42:27.927 libsrtp Ugh: /dev/urandom not present, using rand() instead 11:42:27.928 libsrtp Ugh: /dev/urandom not present, using rand() instead 11:42:27.929 sip_endpoint.c Module "mod-evsub" registered 11:42:27.929 sip_endpoint.c Module "mod-presence" registered 11:42:27.929 sip_endpoint.c Module "mod-refer" registered 11:42:27.929 sip_endpoint.c Module "mod-pjsua-pres" registered 11:42:27.929 sip_endpoint.c Module "mod-pjsua-im" registered 11:42:27.929 sip_endpoint.c Module "mod-pjsua-options" registered 11:42:27.930 pjsua_core.c 1 SIP worker threads created 11:42:27.930 pjsua_core.c pjsua version 1.3 for i686-pc-mingw32 initialized 11:42:27.931 pjsua_core.c SIP UDP socket reachable at 192.168.1.100:5060 11:42:27.931 udp0x1b5f0a8 SIP UDP transport started, published address is 192.168.1.100:5060 11:42:27.932 pjsua_media.c RTP socket reachable at 192.168.1.100:4000 11:42:27.932 pjsua_media.c RTCP socket reachable at 192.168.1.100:4001 11:42:27.933 pjsua_media.c RTP socket reachable at 192.168.1.100:4002 11:42:27.933 pjsua_media.c RTCP socket reachable at 192.168.1.100:4003 11:42:27.934 pjsua_media.c RTP socket reachable at 192.168.1.100:4004 11:42:27.934 pjsua_media.c RTCP socket reachable at 192.168.1.100:4005 11:42:27.937 pjsua_media.c RTP socket reachable at 192.168.1.100:4006 11:42:27.937 pjsua_media.c RTCP socket reachable at 192.168.1.100:4007 11:42:27.937 pjsua_acc.c Account sip:1111 at 192.168.1.106 added with id 0 11:42:27.938 pjsua_core.c TX 386 bytes Request msg REGISTER/cseq=34513 (tdta0x1b77838) to UDP 192.168.1.106:5060: REGISTER sip:192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjcdb582acdca44a1290bf39935c2a2d3f Max-Forwards: 70 From: <sip:1111@192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c To: <sip:1111 at 192.168.1.106> Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9 CSeq: 34513 REGISTER Contact: <sip:1111 at 192.168.1.100:5060> Expires: 300 Content-Length: 0 --end msg-- 11:42:27.938 pjsua_acc.c Registration sent 11:42:27.938 pjsua_media.c Opening sound device PCM at 16000/1/20ms 11:42:27.943 pjsua_core.c RX 561 bytes Response msg 401/REGISTER/cseq=34513 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjcdb582acdca44a1290bf39935c2a2d3f;received=192.168.1.100;rport=5060 From: <sip:1111@192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c To: <sip:1111 at 192.168.1.106>;tag=as6d46783e Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9 CSeq: 34513 REGISTER User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33aa239f" Content-Length: 0 --end msg-- 11:42:27.944 pjsua_core.c TX 546 bytes Request msg REGISTER/cseq=34514 (tdta0x1b77838) to UDP 192.168.1.106:5060: REGISTER sip:192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjbc6c1075678a468191ad325e6d9ae705 Max-Forwards: 70 From: <sip:1111@192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c To: <sip:1111 at 192.168.1.106> Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9 CSeq: 34514 REGISTER Contact: <sip:1111 at 192.168.1.100:5060> Expires: 300 Authorization: Digest username="1111", realm="asterisk", nonce="33aa239f", uri="sip:192.168.1.106", response="1722c88c9aa369ec0fc019b0639face8", algorithm=MD5 Content-Length: 0 --end msg-- 11:42:27.947 pjsua_core.c RX 535 bytes Request msg OPTIONS/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060: OPTIONS sip:1111 at 192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK0452b14f;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@192.168.1.106>;tag=as6611d81e To: <sip:1111 at 192.168.1.100:5060> Contact: <sip:Unknown at 192.168.1.106> Call-ID: 1c1f29b871e9de0a1353986b74b4a081 at 192.168.1.106 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Date: Wed, 19 Aug 2009 04:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --end msg-- 11:42:28.008 ec0x1b7dcb0 AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms 11:42:28.008 pjsua_call.c Making call with acc #0 to sip:2222 at 192.168.1.106 11:42:28.009 pjsua_media.c Media index 0 selected for call 0 11:42:28.009 pjsua_core.c TX 998 bytes Request msg INVITE/cseq=17311 (tdta0x1bade88) to UDP 192.168.1.106:5060: INVITE sip:2222 at 192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300 Max-Forwards: 70 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106 Contact: <sip:1111 at 192.168.1.100:5060> Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17311 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length: 462 v=0 o=- 3459670948 3459670948 IN IP4 192.168.1.100 s=pjmedia c=IN IP4 192.168.1.100 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.1.100 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 11:42:28.009 APP Call 0 state=CALLING Press 'h' to hangup all calls, 'q' to quit 11:42:28.009 pjsua_core.c TX 1090 bytes Response msg 200/OPTIONS/cseq=102 (tdta0x1bb0640) to UDP 192.168.1.106:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK0452b14f Call-ID: 1c1f29b871e9de0a1353986b74b4a081 at 192.168.1.106 From: "Unknown" <sip:Unknown@192.168.1.106>;tag=as6611d81e To: <sip:1111 at 192.168.1.100> CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, norefersub Allow-Events: presence, refer Content-Type: application/sdp Content-Length: 451 v=0 o=- 3459670948 3459670948 IN IP4 192.168.1.100 s=pjmedia c=IN IP4 192.168.1.100 t=0 0 m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.1.100 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 11:42:28.010 pjsua_core.c RX 578 bytes Response msg 200/REGISTER/cseq=34514 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjbc6c1075678a468191ad325e6d9ae705;received=192.168.1.100;rport=5060 From: <sip:1111@192.168.1.106>;tag=94ee42002adc4cf2894281cb11da935c To: <sip:1111 at 192.168.1.106>;tag=as6d46783e Call-ID: 02e3fa1a88cf4317a99db0d69b5a48e9 CSeq: 34514 REGISTER User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 300 Contact: <sip:1111 at 192.168.1.100:5060>;expires=300 Date: Wed, 19 Aug 2009 04:42:52 GMT Content-Length: 0 --end msg-- 11:42:28.010 pjsua_acc.c sip:1111 at 192.168.1.106: registration success, status=200 (OK), will re-register in 300 seconds 11:42:28.010 pjsua_acc.c Keep-alive timer started for acc 0, destination:192.168.1.106:5060, interval:15s 11:42:28.012 pjsua_core.c RX 555 bytes Response msg 401/INVITE/cseq=17311 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300;received=192.168.1.100;rport=5060 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as779078f4 Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17311 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23761d3e" Content-Length: 0 --end msg-- 11:42:28.012 pjsua_core.c TX 338 bytes Request msg ACK/cseq=17311 (tdta0x1bb0640) to UDP 192.168.1.106:5060: ACK sip:2222 at 192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjb3b0682c5e5d4575b603d34f615f2300 Max-Forwards: 70 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as779078f4 Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17311 ACK Content-Length: 0 --end msg-- 11:42:28.012 pjsua_core.c TX 1163 bytes Request msg INVITE/cseq=17312 (tdta0x1bade88) to UDP 192.168.1.106:5060: INVITE sip:2222 at 192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c Max-Forwards: 70 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106 Contact: <sip:1111 at 192.168.1.100:5060> Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Authorization: Digest username="1111", realm="asterisk", nonce="23761d3e", uri="sip:2222 at 192.168.1.106", response="b504983e43d6d256f3e46f1b2744c28a", algorithm=MD5 Content-Type: application/sdp Content-Length: 462 v=0 o=- 3459670948 3459670948 IN IP4 192.168.1.100 s=pjmedia c=IN IP4 192.168.1.100 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101 a=rtcp:4001 IN IP4 192.168.1.100 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 11:42:28.013 APP Call 0 state=CALLING 11:42:28.018 pjsua_core.c RX 493 bytes Response msg 100/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106 Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2222 at 192.168.1.106> Content-Length: 0 --end msg-- 11:42:28.539 pjsua_core.c RX 509 bytes Response msg 180/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as7b91226b Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2222 at 192.168.1.106> Content-Length: 0 --end msg-- 11:42:28.539 APP Call 0 state=EARLY 11:42:28.578 pjsua_core.c RX 509 bytes Response msg 180/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as7b91226b Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2222 at 192.168.1.106> Content-Length: 0 --end msg-- 11:42:28.578 APP Call 0 state=EARLY 11:42:32.481 pjsua_core.c RX 835 bytes Response msg 200/INVITE/cseq=17312 (rdata0x1b68864) from UDP 192.168.1.106:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj7f1601f6f7b242b5a1067a1a149ed32c;received=192.168.1.100;rport=5060 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as7b91226b Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2222 at 192.168.1.106> Content-Type: application/sdp Content-Length: 298 v=0 o=root 1075496772 1075496772 IN IP4 192.168.1.106 s=Asterisk PBX 1.6.0.9-samy-r27 c=IN IP4 192.168.1.106 t=0 0 m=audio 11456 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --end msg-- 11:42:32.481 APP Call 0 state=CONNECTING 11:42:32.482 strm0x1bb2f64 VAD temporarily disabled 11:42:32.482 strm0x1bb2f64 Encoder stream started 11:42:32.482 strm0x1bb2f64 Decoder stream started 11:42:32.482 pjsua_media.c Media updates, stream #0: PCMU (sendrecv) 11:42:32.483 conference.c Port 1 (sip:2222 at 192.168.1.106) transmitting to port 0 (Microsoft Sound Mapper - Input) 11:42:32.483 conference.c Port 0 (Microsoft Sound Mapper - Input) transmitting to port 1 (sip:2222 at 192.168.1.106) 11:42:32.483 pjsua_core.c TX 338 bytes Request msg ACK/cseq=17312 (tdta0x1bb6610) to UDP 192.168.1.106:5060: ACK sip:2222 at 192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPjf34ed215c1cc43d3b235d7f73b136980 Max-Forwards: 70 From: sip:1111@192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 To: sip:2222 at 192.168.1.106;tag=as7b91226b Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 17312 ACK Content-Length: 0 --end msg-- 11:42:32.483 APP Call 0 state=CONFIRMED 11:42:32.485 Master/sound Underflow, buf_cnt=0, will generate 1 frame 11:42:33.116 strm0x1bb2f64 VAD re-enabled 11:42:40.434 ec0x1b7dcb0 Underflow, buf_cnt=0, will generate 1 frame 11:42:43.352 pjsua_core.c RX 430 bytes Request msg BYE/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060: BYE sip:1111 at 192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK591a44dd;rport Max-Forwards: 70 From: sip:2222@192.168.1.106;tag=as7b91226b To: sip:1111 at 192.168.1.106;tag=27b9f31d7d084a55ac6daff93aa6f342 Call-ID: 74fac184beef419eb84af5e932b8a8e5 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --end msg-- 11:42:43.352 pjsua_core.c TX 304 bytes Response msg 200/BYE/cseq=102 (tdta0x1b77838) to UDP 192.168.1.106:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK591a44dd Call-ID: 74fac184beef419eb84af5e932b8a8e5 From: <sip:2222@192.168.1.106>;tag=as7b91226b To: <sip:1111 at 192.168.1.106>;tag=27b9f31d7d084a55ac6daff93aa6f342 CSeq: 102 BYE Content-Length: 0 --end msg-- 11:42:43.353 APP Call 0 state=DISCONNCTD 11:42:43.353 pjsua_media.c Media session for call 0 is destroyed 11:42:44.353 pjsua_media.c Closing sound device after idle for 1 seconds 11:42:44.353 pjsua_media.c Closing Microsoft Sound Mapper - Output sound playback device and Microsoft Sound Mapper - Input sound capture device 11:42:58.290 pjsua_core.c RX 954 bytes Request msg INVITE/cseq=102 (rdata0x1b68864) from UDP 192.168.1.106:5060: INVITE sip:1111 at 192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK69042457;rport Max-Forwards: 70 From: "2222" <sip:2222@192.168.1.106>;tag=as37bba9bb To: <sip:1111 at 192.168.1.100:5060> Contact: <sip:2222 at 192.168.1.106> Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9-samy-r27 Date: Wed, 19 Aug 2009 04:43:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 397 v=0 o=root 2008801089 2008801089 IN IP4 192.168.1.106 s=Asterisk PBX 1.6.0.9-samy-r27 c=IN IP4 192.168.1.106 b=CT:384 t=0 0 m=audio 10282 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 17420 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 a=sendrecv --end msg-- 11:42:58.291 pjsua_media.c Media index 0 selected for call 1 11:42:58.291 pjsua_core.c TX 295 bytes Response msg 100/INVITE/cseq=102 (tdta0x1bb3a80) to UDP 192.168.1.106:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK69042457 Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106 From: "2222" <sip:2222@192.168.1.106>;tag=as37bba9bb To: <sip:1111 at 192.168.1.100> CSeq: 102 INVITE Content-Length: 0 --end msg-- 11:42:58.292 strm0x1b7a15c VAD temporarily disabled 11:42:58.292 strm0x1b7a15c Encoder stream started 11:42:58.293 strm0x1b7a15c Decoder stream started 11:42:58.293 pjsua_media.c Media updates, stream #0: PCMU (sendrecv) 11:42:58.293 pjsua_media.c Opening sound device PCM at 16000/1/20ms 11:42:58.398 ec0x1bb5498 AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms 11:42:58.399 conference.c Port 1 (sip:2222 at 192.168.1.106) transmitting to port 0 (Microsoft Sound Mapper - Input) 11:42:58.399 conference.c Port 0 (Microsoft Sound Mapper - Input) transmitting to port 1 (sip:2222 at 192.168.1.106) 11:42:58.399 pjsua_core.c TX 862 bytes Response msg 200/INVITE/cseq=102 (tdta0x1bb3a80) to UDP 192.168.1.106:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:5060;rport=5060;received=192.168.1.106;branch=z9hG4bK69042457 Call-ID: 4a3bf73b6bd880733eeb2c8a6475dca2 at 192.168.1.106 From: "2222" <sip:2222@192.168.1.106>;tag=as37bba9bb To: <sip:1111 at 192.168.1.100>;tag=ad4e2b7224ed4585a3ba37680f8f19f6 CSeq: 102 INVITE Contact: <sip:1111 at 192.168.1.100:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length: 327 v=0 o=- 3459670978 3459670979 IN IP4 192.168.1.100 s=pjmedia c=IN IP4 192.168.1.100 t=0 0
