Benny, Attached?are?the log files a1_1.log and a1_2.log for the cases 1 and 2 in my previous email. ? I run the pjsua on a PC with Window XP. The softphone is run on a PocketPC with Window Mobile 6. I use a wireless router of LinkSys WRT54GLa to handle the network traffic.?The PocketPC talks with the network using 800.11g. The phone calls are SIP direct call without a proxy server. To run pjsua, there is no other argument except "--log-file=a1.log --log-level=6". ? The IP address of PC and PocketPC?are Ip1 and??Ip2 respectively. ? 1. PocketPC calls pjsua Issue calls from the softphone to "sip:3000 at Ip1". To answer the call on pjsua, just type command "a" and then "222". I can hear sounds from the PocketPC side but not the pjsua side. Using command "dq", I found that very small number of RTP packets reach at the pjsua. ? 2. pjsua calls PocketPC Issue calls using command "m" on pjsua, then with "sip:3001 at Ip2". I can not hear sounds on both sides, although they are connected. ? Please let me know if you want more information. THanks. Paul == a1_1.log 17:12:48.481 sip_endpoint.c Module "mod-pjsua-log" registered ?17:12:48.481 sip_endpoint.c Module "mod-tsx-layer" registered ?17:12:48.481 sip_endpoint.c Module "mod-stateful-util" registered ?17:12:48.481 sip_endpoint.c Module "mod-ua" registered ?17:12:48.481 sip_endpoint.c Module "mod-100rel" registered ?17:12:48.481 sip_endpoint.c Module "mod-pjsua" registered ?17:12:48.481 sip_endpoint.c Module "mod-invite" registered ?17:12:48.481????? pasound.c PA message: before paHostApiInitializers[0]. ?17:12:48.528????? pasound.c PA message: Pa_GetDeviceInfo: Num input channels reported as 65535! Changed to 2. ?17:12:48.528????? pasound.c PA message: Pa_GetDeviceInfo: Num output channels reported as 65535! Changed to 2. ?17:12:48.528????? pasound.c PA message: after paHostApiInitializers[0]. ?17:12:48.528????? pasound.c PA message: before paHostApiInitializers[1]. ?17:12:48.590????? pasound.c PA message: after paHostApiInitializers[1]. ?17:12:48.590????? pasound.c PA message: before paHostApiInitializers[2]. ?17:12:48.590????? pasound.c PA message: after paHostApiInitializers[2]. ?17:12:48.590????? pasound.c PortAudio sound library initialized, status=0 ?17:12:48.590????? pasound.c PortAudio host api count=3 ?17:12:48.590????? pasound.c Sound device count=8 ?17:12:48.590????????? pjlib select() I/O Queue created (003CB03C) ?17:12:48.590?? conference.c Creating conference bridge with 254 ports ?17:12:48.590?? conference.c Sound device successfully created for port 0 ?17:12:48.590 sip_endpoint.c Module "mod-evsub" registered ?17:12:48.590 sip_endpoint.c Module "mod-presence" registered ?17:12:48.590??????? evsub.c Event pkg "presence" registered by mod-presence ?17:12:48.590 sip_endpoint.c Module "mod-refer" registered ?17:12:48.590??????? evsub.c Event pkg "refer" registered by mod-refer ?17:12:48.590 sip_endpoint.c Module "mod-pjsua-pres" registered ?17:12:48.590 sip_endpoint.c Module "mod-pjsua-im" registered ?17:12:48.590 sip_endpoint.c Module "mod-pjsua-options" registered ?17:12:48.590?? pjsua_core.c 1 SIP worker threads created ?17:12:48.590?? pjsua_core.c pjsua version 0.8.0-trunk for win32 initialized ?17:12:48.590?? pjsua_core.c SIP UDP socket reachable at Ip3:5060 ?17:12:48.590??? udp00DBD7D8 SIP UDP transport started, published address is Ip3:5060 ?17:12:48.590??? pjsua_acc.c Account <sip:Ip3:5060> added with id 0 ?17:12:48.590??? tcplis:5060 SIP TCP listener ready for incoming connections at Ip3:5060 ?17:12:48.590??? pjsua_acc.c Account <sip:Ip3:5060;transport=TCP> added with id 1 ?17:12:48.606? pjsua_media.c RTP socket reachable at Ip3:4000 ?17:12:48.606? pjsua_media.c RTCP socket reachable at Ip3:4001 ?17:12:48.606? pjsua_media.c RTP socket reachable at Ip3:4002 ?17:12:48.606? pjsua_media.c RTCP socket reachable at Ip3:4003 ?17:12:48.606? pjsua_media.c RTP socket reachable at Ip3:4004 ?17:12:48.606? pjsua_media.c RTCP socket reachable at Ip3:4005 ?17:12:48.606? pjsua_media.c RTP socket reachable at Ip3:4006 ?17:12:48.606? pjsua_media.c RTCP socket reachable at Ip3:4007 ?17:12:48.606? pjsua_media.c pjsua_set_snd_dev(): attempting to open devices @16000 Hz ?17:12:48.606????? pasound.c Opened device Conexant HD Audio input(MME)/Conexant HD Audio output(MME) for recording and playback, sample rate=16000, ch=1, bits=16, 320 samples per frame, input latency=100 ms, output latency=100 ms ?17:12:48.606????? pasound..c Starting Conexant HD Audio input stream.. ?17:12:48.606????? pasound.c PA message: Pa_StartStream: waveInStart returned = 0x0. ?17:12:48.606????? pasound.c Done, status=0 ?17:12:48.622?? echo_speex.c Speex Echo canceller/AEC created, clock_rate=16000, samples per frame=320, tail length=200 ms, latency=200 ms ?17:12:48.637????? pasound.c Player thread started ?17:12:48.637????? pasound.c Recorder thread started ?17:12:48.637??? aec00DCC2B0 AEC reset, delay=0, prefetch=4 ?17:12:48.637??? aec00DCC2B0 AEC reset, delay=0, prefetch=4 ?17:12:48.668??? aec00DCC2B0? AEC Info: old frame removed (seq=2, want=-2, count=1) ?17:12:48.668??? aec00DCC2B0? AEC Info: empty queue for seq=-2! ?17:12:48.684??? aec00DCC2B0? AEC Info: old frame removed (seq=3, want=-1, count=1) ?17:12:48.684??? aec00DCC2B0? AEC Info: empty queue for seq=-1! ?17:12:48.700??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:12:48.731??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:12:48.747??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:12:48.762??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:12:53.637?? sound_port.c EC suspended because of inactivity ?17:12:57.247 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=1 (rdata00DBDC9C) ?17:12:57.247?? pjsua_core.c RX 705 bytes Request msg INVITE/cseq=1 (rdata00DBDC9C) from UDP Ip2:5060: INVITE sip:3000 at Ip1 SIP/2.0 To: <sip:3000 at Ip1> From: "PPC-3001"<sip:Ip2>;tag=2807035031951 Via: SIP/2.0/UDP Ip2;rport;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 CSeq: 1 INVITE Contact: <sip:Ip2> Max-Forwards: 70 User-Agent: SJphone/1.60.303c (SJ Labs) Content-Length: 268 Content-Type: application/sdp v=0 o=- 3422131971 3422131971 IN IP4 Ip2 s=SJphone c=IN IP4 Ip2 t=0 0 a=direction:active m=audio 49222 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --end msg-- ?17:12:57.247??? tsx00E006C4 Transaction created for Request msg INVITE/cseq=1 (rdata00DBDC9C) ?17:12:57.247??? tsx00E006C4 Incoming Request msg INVITE/cseq=1 (rdata00DBDC9C) in state Null ?17:12:57.247??? tsx00E006C4 State changed from Null to Trying, event=RX_MSG ?17:12:57.247??? dlg00DFFE94 Transaction tsx00E006C4 state changed to Trying ?17:12:57.247??? dlg00DFFE94 UAS dialog created ?17:12:57.247??? dlg00DFFE94 Module mod-invite added as dialog usage, data=00E0050C ?17:12:57.247??? dlg00DFFE94 Session count inc to 2 by mod-invite ?17:12:57.247??? inv00DFFE94 UAS invite session created for dialog dlg00DFFE94 ?17:12:57.247?? pjsua_call.c Call 0: remote NAT type is 0 (Unknown) ?17:12:57.247?????? endpoint Response msg 100/INVITE/cseq=1 (tdta00E02810) created ?17:12:57.247??? dlg00DFFE94 Initial answer Response msg 100/INVITE/cseq=1 (tdta00E02810) ?17:12:57.247??? inv00DFFE94 Sending Response msg 100/INVITE/cseq=1 (tdta00E02810) ?17:12:57.247??? dlg00DFFE94 Sending Response msg 100/INVITE/cseq=1 (tdta00E02810) ?17:12:57.247??? tsx00E006C4 Sending Response msg 100/INVITE/cseq=1 (tdta00E02810) in state Trying ?17:12:57.247?? pjsua_core.c TX 326 bytes Response msg 100/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1> CSeq: 1 INVITE Content-Length:? 0 --end msg-- ?17:12:57.247??? tsx00E006C4 State changed from Trying to Proceeding, event=TX_MSG ?17:12:57.247??? dlg00DFFE94 Transaction tsx00E006C4 state changed to Proceeding ?17:12:57.247??? pjsua_app.c Call 0 state changed to INCOMING ?17:12:57.262??? pjsua_app.c Incoming call for account 1! From: "PPC-3001" <sip:Ip2> To: <sip:3000 at Ip1> Press a to answer or h to reject call ?17:13:00.372??? inv00DFFE94 SDP negotiation done, status=0 ?17:13:00.372?? pjsua_call.c Call 0: remote NAT type is 0 (Unknown) ?17:13:00.372?? strm00E05724 VAD temporarily disabled ?17:13:00.372????????? rtp.c pjmedia_rtp_session_init: ses=00E06888, default_pt=8, ssrc=0x294823 ?17:13:00.372????????? rtp.c pjmedia_rtp_session_init: ses=00E07490, default_pt=8, ssrc=0x294823 ?17:13:00.372?????? stream.c Stream strm00E05724 created ?17:13:00.387?? strm00E05724 Encoder stream started ?17:13:00.387?? strm00E05724 Decoder stream started ?17:13:00.387???? resample.c resample created: high qualiy, large filter, in/out rate=8000/16000 ?17:13:00.387???? resample.c resample created: high qualiy, large filter, in/out rate=16000/8000 ?17:13:00.387? pjsua_media.c Media updates, stream #0: PCMA (sendrecv) ?17:13:00.387?? conference.c Port 1 (sip:Ip2) transmitting to port 0 (Conexant HD Audio input) ?17:13:00.387?? conference.c Port 0 (Conexant HD Audio input) transmitting to port 1 (sip:Ip2) ?17:13:00.387??? pjsua_app.c Media for call 0 is active ?17:13:00.387??? inv00DFFE94 Sending Response msg 222/INVITE/cseq=1 (tdta00E02810) ?17:13:00.387??? dlg00DFFE94 Sending Response msg 222/INVITE/cseq=1 (tdta00E02810) ?17:13:00.387??? tsx00E006C4 Sending Response msg 222/INVITE/cseq=1 (tdta00E02810) in state Proceeding ?17:13:00.387?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:00.387??? tsx00E006C4 State changed from Proceeding to Completed, event=TX_MSG ?17:13:00.387??? dlg00DFFE94 Transaction tsx00E006C4 state changed to Completed ?17:13:00.387??? pjsua_app.c Call 0 state changed to CONNECTING ?17:13:00.403?? strm00E05724 Jitter buffer empty (prefetch=1) ?17:13:00.403?? strm00E05724 Start talksprut.. ?17:13:00.887??? tsx00E006C4 Retransmit timer event ?17:13:00.887??? tsx00E006C4 Retransmiting Response msg 222/INVITE/cseq=1 (tdta00E02810), count=0, restart?=1 ?17:13:00.887?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:01.012?? strm00E05724 VAD re-enabled ?17:13:01.887??? tsx00E006C4 Retransmit timer event ?17:13:01.887??? tsx00E006C4 Retransmiting Response msg 222/INVITE/cseq=1 (tdta00E02810), count=1, restart?=1 ?17:13:01.887?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:03.012?? silencedet.c Vad cur_threshold updated 4-->41. Signal lo=77 ?17:13:03.840?? Master/sound Overflow, 479 samples reduced, buf_cnt=801 ?17:13:03.840?? Master/sound Buffer size adjusted from 1280 to 801 ?17:13:03.887??? tsx00E006C4 Retransmit timer event ?17:13:03.887??? tsx00E006C4 Retransmiting Response msg 222/INVITE/cseq=1 (tdta00E02810), count=2, restart?=1 ?17:13:03.887?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:05.043?? silencedet.c Vad cur_threshold updated 41-->45. Signal lo=48 ?17:13:06.356?? Master/sound Overflow, 193 samples reduced, buf_cnt=928 ?17:13:06.356?? Master/sound Buffer size adjusted from 1121 to 928 ?17:13:07.043?? silencedet.c Vad cur_threshold updated 45-->47. Signal lo=48 ?17:13:07.059?? Master/sound Overflow, 160 samples reduced, buf_cnt=1088 ?17:13:07.059?? Master/sound Buffer size adjusted from 1248 to 1088 ?17:13:07.887??? tsx00E006C4 Retransmit timer event ?17:13:07.887??? tsx00E006C4 Retransmiting Response msg 222/INVITE/cseq=1 (tdta00E02810), count=3, restart?=1 ?17:13:07.887?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:08.340?? Master/sound Overflow, 302 samples reduced, buf_cnt=786 ?17:13:08.340?? Master/sound Buffer size adjusted from 1088 to 786 ?17:13:08.653?? Master/sound Overflow, 274 samples reduced, buf_cnt=832 ?17:13:08.653?? Master/sound Buffer size adjusted from 1106 to 832 ?17:13:09.059?? silencedet.c Vad cur_threshold updated 47-->58. Signal lo=69 ?17:13:10.247?? Master/sound Overflow, 366 samples reduced, buf_cnt=786 ?17:13:10.247?? Master/sound Buffer size adjusted from 1152 to 786 ?17:13:10.887??? pjsua_app.c ? [CONNECTING] To: "PPC-3001" <sip:Ip2>;tag=2807035031951 ??? Call time: 00h:00m:00s, 1st res in 3125 ms, conn in 0ms ??? SRTP status: Not active Crypto-suite: (null) ??? #0 PCMA @8KHz, sendrecv, peer=Ip2:49222 ?????? RX pt=8, stat last update: never ????????? total 1pkt 0B (40B +IP hdr) @avg=0bps/30bps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?????? TX pt=8, ptime=20ms, stat last update: never ????????? total 525pkt 84.0KB (105.0KB +IP hdr) @avg 63.9Kbps/79.8Kbps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????? RTT msec?????? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?17:13:11.059?? silencedet.c Vad cur_threshold updated 58-->67. Signal lo=76 ?17:13:11.840?? Master/sound Overflow, 319 samples reduced, buf_cnt=787 ?17:13:11.840?? Master/sound Buffer size adjusted from 1106 to 787 ?17:13:11.887??? tsx00E006C4 Retransmit timer event ?17:13:11.887??? tsx00E006C4 Retransmiting Response msg 222/INVITE/cseq=1 (tdta00E02810), count=4, restart?=1 ?17:13:11.887?? pjsua_core.c TX 848 bytes Response msg 222/INVITE/cseq=1 (tdta00E02810) to UDP Ip2:5060: SIP/2.0 222 Default status message Via: SIP/2.0/UDP Ip2;rport=5060;received=Ip2;branch=z9hG4bKc0a8016600000118484f188300003f6a00000176 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 From: "PPC-3001" <sip:Ip2>;tag=2807035031951 To: <sip:3000 at Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a CSeq: 1 INVITE Contact: <sip:Ip3:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length:?? 257 v=0 o=- 3422106777 3422106778 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 8 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?17:13:13.075?? silencedet.c Vad cur_threshold updated 67-->73. Signal lo=78 ?17:13:13.434?? Master/sound Overflow, 322 samples reduced, buf_cnt=785 ?17:13:13.434?? Master/sound Buffer size adjusted from 1107 to 785 ?17:13:13.684????????? pjsua Thread stack max usage=12725 by d:\people\siu\projects\voip\pjsip\pjlib\src\pj\os_core_win32.c:1380 ?17:13:13.684?????? endpoint Request msg BYE/cseq=9210 (tdta00DFDF80) created. ?17:13:13.684??? inv00DFFE94 Sending Request msg BYE/cseq=9210 (tdta00DFDF80) ?17:13:13.684??? dlg00DFFE94 Sending Request msg BYE/cseq=9210 (tdta00DFDF80) ?17:13:13.684??? tsx00DFF014 Transaction created for Request msg BYE/cseq=9209 (tdta00DFDF80) ?17:13:13.684??? tsx00DFF014 Sending Request msg BYE/cseq=9209 (tdta00DFDF80) in state Null ?17:13:13.684? sip_resolve.c Target 'Ip2:0' type=Unspecified resolved to 'Ip2:5060' type=UDP (UDP transport) ?17:13:13.684?? pjsua_core.c TX 402 bytes Request msg BYE/cseq=9209 (tdta00DFDF80) to UDP Ip2:5060: BYE sip:Ip2 SIP/2.0 Via: SIP/2.0/UDP Ip3:5060;rport;branch=z9hG4bKPj5a37dd8329064baf9442a1f3586f8f1f Max-Forwards: 70 From: <sip:3000@Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a To: "PPC-3001" <sip:Ip2>;tag=2807035031951 Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 CSeq: 9209 BYE User-Agent: PJSUA v0.8.0-trunk/win32 Content-Length:? 0 --end msg-- ?17:13:13.684??? tsx00DFF014 State changed from Null to Calling, event=TX_MSG ?17:13:13.684??? dlg00DFFE94 Transaction tsx00DFF014 state changed to Calling ?17:13:13.684?? pjsua_core.c Shutting down... ?17:13:14.075 sip_endpoint.c Processing incoming message: Response msg 200/BYE/cseq=9209 (rdata00DBDC9C) ?17:13:14.075?? pjsua_core.c RX 394 bytes Response msg 200/BYE/cseq=9209 (rdata00DBDC9C) from UDP Ip2:5060: SIP/2.0 200 OK To: "PPC-3001"<sip:Ip2>;tag=2807035031951 From: <sip:3000@Ip1>;tag=c4dc7d86ffd44336a9d179b7d43ff57a Via: SIP/2.0/UDP Ip3:5060;rport=5060;received=Ip1;branch=z9hG4bKPj5a37dd8329064baf9442a1f3586f8f1f Call-ID: 00002716-7A65-0000-3459-0000C67B0000 at Ip2 CSeq: 9209 BYE Content-Length: 0 Server: SJphone/1.60.303c (SJ Labs) --end msg-- ?17:13:14.075??? tsx00DFF014 Incoming Response msg 200/BYE/cseq=9209 (rdata00DBDC9C) in state Calling ?17:13:14.075??? tsx00DFF014 State changed from Calling to Completed, event=RX_MSG ?17:13:14.075??? dlg00DFFE94 Received Response msg 200/BYE/cseq=9209 (rdata00DBDC9C) ?17:13:14.075??? dlg00DFFE94 Transaction tsx00DFF014 state changed to Completed ?17:13:14.075??? pjsua_app.c Call 0 is DISCONNECTED [reason=603 (Decline)] ?17:13:14.075??? pjsua_app.c Call 0 disconnected, dumping media stats ? [DISCONNCTD] To: "PPC-3001" <sip:Ip2>;tag=2807035031951 ??? Call time: 00h:00m:00s, 1st res in 3125 ms, conn in 0ms ??? SRTP status: Not active Crypto-suite: (null) ??? #0 PCMA @8KHz, sendrecv, peer=Ip2:49222 ?????? RX pt=8, stat last update: 00h:00m:01.922s ago ????????? total 1pkt 0B (40B +IP hdr) @avg=0bps/23bps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?????? TX pt=8, ptime=20ms, stat last update: never ????????? total 684pkt 109.4KB (136.8KB +IP hdr) @avg 63.8Kbps/79.8Kbps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????? RTT msec?????? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?17:13:14.075? pjsua_media.c Media session for call 0 is destroyed ?17:13:14.075??? dlg00DFFE94 Session count dec to 1 by mod-invite ?17:13:14.684? pjsua_media.c Closing (null) sound playback device and (null) sound capture device ?17:13:14.715????? pasound.c Stopping stream.. ?17:13:14.715????? pasound.c PA message: WinMME StopStream: waiting for background thread. ?17:13:14.793????? pasound.c PA message: WinMME StopStream: waiting for background thread. ?17:13:14.809????? pasound.c Done, status=0 ?17:13:14.809????? pasound.c Closing Conexant HD Audio input: 0 underflow, 0 overflow ?17:13:15.309????????? media Thread stack max usage=2165 by d:\people\siu\projects\voip\pjsip\pjlib\src\pj\os_core_win32.c:903 ?17:13:15.309????? pasound.c PortAudio sound library shutting down.. ?17:13:15.309????? pasound..c PA message: TerminateHostApis in ?17:13:15.309????? pasound.c PA message: TerminateHostApis out ?17:13:15.309 sip_endpoint.c Destroying endpoing instance.. ?17:13:15.309 sip_endpoint.c Module "mod-pjsua-options" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-pjsua-im" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-pjsua-pres" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-pjsua" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-stateful-util" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-refer" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-presence" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-evsub" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-invite" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-100rel" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-ua" unregistered ?17:13:15.309 sip_transactio Stopping transaction layer module ?17:13:15.309?? tdta00DFDF80 Destroying txdata Request msg BYE/cseq=9209 (tdta00DFDF80) ?17:13:15.309??? tsx00DFF014 Transaction destroyed! ?17:13:15.309??? tsx00E006C4 Transaction destroyed! ?17:13:15.309 sip_transactio Transaction layer module destroyed ?17:13:15.309 sip_endpoint.c Module "mod-tsx-layer" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-msg-print" unregistered ?17:13:15.309 sip_endpoint.c Module "mod-pjsua-log" unregistered ?17:13:15.309 sip_transport. Destroying transport manager ?17:13:15.309??? tcplis:5060 SIP TCP listener destroyed ?17:13:15.309 sip_transport. Warning: 1 transmit buffer(s) not freed! ?17:13:15.309 sip_endpoint.c Endpoint 003C6BFC destroyed ?17:13:15.309??? dlg00DFFE30 Pool is not released by application, releasing now ?17:13:15.309?? tdta00E02810 Pool is not released by application, releasing now ?17:13:15.309?? pjsua_core.c PJSUA destroyed... == == a1_2.log ?17:11:56.653 sip_endpoint.c Module "mod-pjsua-log" registered ?17:11:56.653 sip_endpoint.c Module "mod-tsx-layer" registered ?17:11:56.653 sip_endpoint.c Module "mod-stateful-util" registered ?17:11:56.653 sip_endpoint.c Module "mod-ua" registered ?17:11:56.653 sip_endpoint.c Module "mod-100rel" registered ?17:11:56.653 sip_endpoint.c Module "mod-pjsua" registered ?17:11:56.653 sip_endpoint.c Module "mod-invite" registered ?17:11:56.653????? pasound.c PA message: before paHostApiInitializers[0]. ?17:11:56.700????? pasound.c PA message: Pa_GetDeviceInfo: Num input channels reported as 65535! Changed to 2. ?17:11:56.700????? pasound.c PA message: Pa_GetDeviceInfo: Num output channels reported as 65535! Changed to 2. ?17:11:56.700????? pasound.c PA message: after paHostApiInitializers[0]. ?17:11:56.700????? pasound.c PA message: before paHostApiInitializers[1]. ?17:11:56.762????? pasound.c PA message: after paHostApiInitializers[1]. ?17:11:56.762????? pasound.c PA message: before paHostApiInitializers[2]. ?17:11:56.762????? pasound.c PA message: after paHostApiInitializers[2]. ?17:11:56.762????? pasound.c PortAudio sound library initialized, status=0 ?17:11:56.762????? pasound.c PortAudio host api count=3 ?17:11:56.762????? pasound.c Sound device count=8 ?17:11:56.762????????? pjlib select() I/O Queue created (003CB03C) ?17:11:56.762?? conference.c Creating conference bridge with 254 ports ?17:11:56.762?? conference.c Sound device successfully created for port 0 ?17:11:56.762 sip_endpoint.c Module "mod-evsub" registered ?17:11:56.762 sip_endpoint.c Module "mod-presence" registered ?17:11:56.762??????? evsub.c Event pkg "presence" registered by mod-presence ?17:11:56.762 sip_endpoint.c Module "mod-refer" registered ?17:11:56.762??????? evsub.c Event pkg "refer" registered by mod-refer ?17:11:56.762 sip_endpoint.c Module "mod-pjsua-pres" registered ?17:11:56.762 sip_endpoint.c Module "mod-pjsua-im" registered ?17:11:56.762 sip_endpoint.c Module "mod-pjsua-options" registered ?17:11:56.762?? pjsua_core.c 1 SIP worker threads created ?17:11:56.762?? pjsua_core.c pjsua version 0.8..0-trunk for win32 initialized ?17:11:56.762?? pjsua_core.c SIP UDP socket reachable at Ip3:5060 ?17:11:56.762??? udp00DBD7D8 SIP UDP transport started, published address is Ip3:5060 ?17:11:56.762??? pjsua_acc.c Account <sip:Ip3:5060> added with id 0 ?17:11:56.762??? tcplis:5060 SIP TCP listener ready for incoming connections at Ip3:5060 ?17:11:56.762??? pjsua_acc.c Account <sip:Ip3:5060;transport=TCP> added with id 1 ?17:11:56.778? pjsua_media.c RTP socket reachable at Ip3:4000 ?17:11:56.778? pjsua_media.c RTCP socket reachable at Ip3:4001 ?17:11:56.778? pjsua_media.c RTP socket reachable at Ip3:4002 ?17:11:56.778? pjsua_media.c RTCP socket reachable at Ip3:4003 ?17:11:56.778? pjsua_media.c RTP socket reachable at Ip3:4004 ?17:11:56.778? pjsua_media.c RTCP socket reachable at Ip3:4005 ?17:11:56.778? pjsua_media.c RTP socket reachable at Ip3:4006 ?17:11:56.778? pjsua_media.c RTCP socket reachable at Ip3:4007 ?17:11:56.778? pjsua_media.c pjsua_set_snd_dev(): attempting to open devices @16000 Hz ?17:11:56.778????? pasound.c Opened device Conexant HD Audio input(MME)/Conexant HD Audio output(MME) for recording and playback, sample rate=16000, ch=1, bits=16, 320 samples per frame, input latency=100 ms, output latency=100 ms ?17:11:56.778????? pasound.c Starting Conexant HD Audio input stream.. ?17:11:56.778????? pasound.c PA message: Pa_StartStream: waveInStart returned = 0x0. ?17:11:56.778????? pasound.c Done, status=0 ?17:11:56.793?? echo_speex.c Speex Echo canceller/AEC created, clock_rate=16000, samples per frame=320, tail length=200 ms, latency=200 ms ?17:11:56.809????? pasound.c Player thread started ?17:11:56.809????? pasound.c Recorder thread started ?17:11:56.809??? aec00DCC2B0 AEC reset, delay=0, prefetch=4 ?17:11:56.809??? aec00DCC2B0 AEC reset, delay=0, prefetch=4 ?17:11:56.840??? aec00DCC2B0? AEC Info: old frame removed (seq=2, want=-2, count=1) ?17:11:56.840??? aec00DCC2B0? AEC Info: empty queue for seq=-2! ?17:11:56.856??? aec00DCC2B0? AEC Info: old frame removed (seq=3, want=-1, count=1) ?17:11:56.856??? aec00DCC2B0? AEC Info: empty queue for seq=-1! ?17:11:56.872??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:11:56.887??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:11:56.918??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:11:56.934??? aec00DCC2B0? AEC Info: prefetching (first seq=4) ?17:12:01.809?? sound_port.c EC suspended because of inactivity ?17:12:12.809?? pjsua_call.c Making call with acc #1 to sip:3001 at Ip2 ?17:12:12.809??? dlg00DFBB3C UAC dialog created ?17:12:12.825??? dlg00DFBB3C Module mod-invite added as dialog usage, data=00DFD940 ?17:12:12.825??? dlg00DFBB3C Session count inc to 2 by mod-invite ?17:12:12.825??? dlg00DFBB3C Module mod-100rel added as dialog usage, data=00DFE820 ?17:12:12.825??? dlg00DFBB3C 100rel module attached ?17:12:12.825??? inv00DFBB3C UAC invite session created for dialog dlg00DFBB3C ?17:12:12..825?????? endpoint Request msg INVITE/cseq=15913 (tdta00DFEA00) created. ?17:12:12.825??? inv00DFBB3C Sending Request msg INVITE/cseq=15913 (tdta00DFEA00) ?17:12:12.825??? dlg00DFBB3C Sending Request msg INVITE/cseq=15913 (tdta00DFEA00) ?17:12:12.825??? tsx00DFFA94 Transaction created for Request msg INVITE/cseq=15912 (tdta00DFEA00) ?17:12:12.825??? tsx00DFFA94 Sending Request msg INVITE/cseq=15912 (tdta00DFEA00) in state Null ?17:12:12.825? sip_resolve.c Target 'Ip2:0' type=Unspecified resolved to 'Ip2:5060' type=UDP (UDP transport) ?17:12:12.825?? pjsua_core.c TX 1043 bytes Request msg INVITE/cseq=15912 (tdta00DFEA00) to UDP Ip2:5060: INVITE sip:3001 at Ip2 SIP/2.0 Via: SIP/2.0/UDP Ip3:5060;rport;branch=z9hG4bKPj66bcf113d71944d989f05c78383a59fa Max-Forwards: 70 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 To: sip:3001 at Ip2 Contact: <sip:Ip3:5060> Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15912 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: PJSUA v0.8.0-trunk/win32 Content-Type: application/sdp Content-Length:?? 465 v=0 o=- 3422106732 3422106732 IN IP4 Ip3 s=pjmedia c=IN IP4 Ip3 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 Ip3 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 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-- ?17:12:12..825??? tsx00DFFA94 State changed from Null to Calling, event=TX_MSG ?17:12:12.825??? dlg00DFBB3C Transaction tsx00DFFA94 state changed to Calling ?17:12:12.825??? pjsua_app.c Call 0 state changed to CALLING ?17:12:13.215 sip_endpoint.c Processing incoming message: Response msg 100/INVITE/cseq=15912 (rdata00DBDC9C) ?17:12:13.215?? pjsua_core.c RX 349 bytes Response msg 100/INVITE/cseq=15912 (rdata00DBDC9C) from UDP Ip2:5060: SIP/2.0 100 Trying To: "Anonymous"<sip:3001 at Ip2>;tag=2802631630558 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 Via: SIP/2.0/UDP Ip3:5060;rport=5060;received=Ip1;branch=z9hG4bKPj66bcf113d71944d989f05c78383a59fa Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15912 INVITE Content-Length: 0 --end msg-- ?17:12:13..215??? tsx00DFFA94 Incoming Response msg 100/INVITE/cseq=15912 (rdata00DBDC9C) in state Calling ?17:12:13.215??? tsx00DFFA94 State changed from Calling to Proceeding, event=RX_MSG ?17:12:13.215??? dlg00DFBB3C Received Response msg 100/INVITE/cseq=15912 (rdata00DBDC9C) ?17:12:13.215??? dlg00DFBB3C Transaction tsx00DFFA94 state changed to Proceeding ?17:12:13.293 sip_endpoint.c Processing incoming message: Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C) ?17:12:13.293?? pjsua_core.c RX 638 bytes Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C) from UDP Ip2:5060: SIP/2.0 200 OK To: "Anonymous"<sip:3001 at Ip2>;tag=2802631630558 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 Via: SIP/2.0/UDP Ip3:5060;rport=5060;received=Ip1;branch=z9hG4bKPj66bcf113d71944d989f05c78383a59fa Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15912 INVITE Contact: <sip:anonymous at Ip2> Content-Length: 220 Content-Type: application/sdp v=0 o=- 3422131927 3422131927 IN IP4 76.235.156.64 s=SJphone c=IN IP4 76.235.156.64 t=0 0 a=direction:active m=audio 49220 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --end msg-- ?17:12:13.293??? tsx00DFFA94 Incoming Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C) in state Proceeding ?17:12:13.293??? tsx00DFFA94 State changed from Proceeding to Terminated, event=RX_MSG ?17:12:13.293??? dlg00DFBB3C Received Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C) ?17:12:13.293??? dlg00DFBB3C Route-set updated ?17:12:13.293??? dlg00DFBB3C Route-set frozen ?17:12:13.293??? dlg00DFBB3C Transaction tsx00DFFA94 state changed to Terminated ?17:12:13.293??? pjsua_app.c Call 0 state changed to CONNECTING ?17:12:13.293??? inv00DFBB3C Got SDP answer in Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C) ?17:12:13.293??? inv00DFBB3C SDP negotiation done, status=0 ?17:12:13.293?? pjsua_call.c Call 0: remote NAT type is 0 (Unknown) ?17:12:13.293?? strm00E03584 VAD temporarily disabled ?17:12:13.293????????? rtp.c pjmedia_rtp_session_init: ses=00E03D6C, default_pt=3, ssrc=0x629e1d02 ?17:12:13.293????????? rtp.c pjmedia_rtp_session_init: ses=00E044F0, default_pt=3, ssrc=0x629e1d02 ?17:12:13.293?????? stream.c Stream strm00E03584 created ?17:12:13.293?? strm00E03584 Encoder stream started ?17:12:13.293?? strm00E03584 Decoder stream started ?17:12:13.293???? resample.c resample created: high qualiy, large filter, in/out rate=8000/16000 ?17:12:13.293???? resample.c resample created: high qualiy, large filter, in/out rate=16000/8000 ?17:12:13.293? pjsua_media.c Media updates, stream #0: GSM (sendrecv) ?17:12:13.293?? conference.c Port 1 (sip:3001 at Ip2) transmitting to port 0 (Conexant HD Audio input) ?17:12:13.293?? conference.c Port 0 (Conexant HD Audio input) transmitting to port 1 (sip:3001 at Ip2) ?17:12:13.293??? pjsua_app.c Media for call 0 is active ?17:12:13.293??? inv00DFBB3C Received Response msg 200/INVITE/cseq=15912 (rdata00DBDC9C), sending ACK ?17:12:13.293?????? endpoint Request msg ACK/cseq=15912 (tdta00E07830) created. ?17:12:13.293??? dlg00DFBB3C Sending Request msg ACK/cseq=15912 (tdta00E07830) ?17:12:13.293? sip_resolve.c Target 'Ip2:0' type=Unspecified resolved to 'Ip2:5060' type=UDP (UDP transport) ?17:12:13.293?? pjsua_core.c TX 345 bytes Request msg ACK/cseq=15912 (tdta00E07830) to UDP Ip2:5060: ACK sip:anonymous at Ip2 SIP/2.0 Via: SIP/2.0/UDP Ip3:5060;rport;branch=z9hG4bKPj5832be8ba85d4c999e93321a97fab48c Max-Forwards: 70 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 To: sip:3001 at Ip2;tag=2802631630558 Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15912 ACK Content-Length:? 0 --end msg-- ?17:12:13.293??? pjsua_app.c Call 0 state changed to CONFIRMED ?17:12:13.293??? tsx00DFFA94 Timeout timer event ?17:12:13.293??? tsx00DFFA94 State changed from Terminated to Destroyed, event=TIMER ?17:12:13.293?? tdta00DFEA00 Destroying txdata Request msg INVITE/cseq=15912 (tdta00DFEA00) ?17:12:13.293??? tsx00DFFA94 Transaction destroyed! ?17:12:13.309?? Master/sound Underflow, buf_cnt=0, will generate 1 frame ?17:12:13.309?? strm00E03584 Jitter buffer empty (prefetch=1) ?17:12:13.309?? strm00E03584 Start talksprut.. ?17:12:13.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.325?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.356?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.372?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.387?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.418?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:13.934?? strm00E03584 VAD re-enabled ?17:12:14.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:15.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:15.950?? silencedet.c Vad cur_threshold updated 4-->41. Signal lo=78 ?17:12:16.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:16.809?? Master/sound Overflow, 205 samples reduced, buf_cnt=1075 ?17:12:16.809?? Master/sound Buffer size adjusted from 1280 to 1075 ?17:12:17.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:17.965?? silencedet.c Vad cur_threshold updated 41-->60. Signal lo=78 ?17:12:18.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:18.403?? Master/sound Overflow, 315 samples reduced, buf_cnt=1080 ?17:12:18.403?? Master/sound Buffer size adjusted from 1395 to 1080 ?17:12:19.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:19.997?? silencedet.c Vad cur_threshold updated 60-->69. Signal lo=78 ?17:12:20.028?? Master/sound Overflow, 479 samples reduced, buf_cnt=921 ?17:12:20..028?? Master/sound Buffer size adjusted from 1400 to 921 ?17:12:20..340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:21.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:21.637?? Master/sound Overflow, 391 samples reduced, buf_cnt=850 ?17:12:21..637?? Master/sound Buffer size adjusted from 1241 to 850 ?17:12:22..012?? silencedet.c Vad cur_threshold updated 69-->74. Signal lo=78 ?17:12:22.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:23.200?? Master/sound Overflow, 160 samples reduced, buf_cnt=1010 ?17:12:23.215?? Master/sound Buffer size adjusted from 1170 to 1010 ?17:12:23.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:24.028?? silencedet.c Vad cur_threshold updated 74-->75. Signal lo=75 ?17:12:24.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:24.809?? Master/sound Overflow, 479 samples reduced, buf_cnt=851 ?17:12:24.809?? Master/sound Buffer size adjusted from 1330 to 851 ?17:12:25.325?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:26.043?? silencedet.c Vad cur_threshold updated 75-->76. Signal lo=76 ?17:12:26.309?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:26.418?? Master/sound Overflow, 354 samples reduced, buf_cnt=817 ?17:12:26.418?? Master/sound Buffer size adjusted from 1171 to 817 ?17:12:27.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:28.012?? Master/sound Overflow, 350 samples reduced, buf_cnt=787 ?17:12:28.012?? Master/sound Buffer size adjusted from 1137 to 787 ?17:12:28.075?? silencedet.c Vad cur_threshold updated 76-->77. Signal lo=78 ?17:12:28.325?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:29.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:29.606?? Master/sound Overflow, 316 samples reduced, buf_cnt=791 ?17:12:29.606?? Master/sound Buffer size adjusted from 1107 to 791 ?17:12:30.090?? silencedet.c Vad cur_threshold updated 77-->78. Signal lo=78 ?17:12:30.340?? strm00E03584 RTP recv() error: Connection reset by peer (WSAECONNRESET) [err:130054] ?17:12:30.747????????? pjsua Thread stack max usage=14013 by d:\people\siu\projects\voip\pjsip\pjlib\src\pj\os_core_win32.c:1380 ?17:12:30.747?????? endpoint Request msg BYE/cseq=15914 (tdta00E08860) created. ?17:12:30.747??? inv00DFBB3C Sending Request msg BYE/cseq=15914 (tdta00E08860) ?17:12:30.747??? dlg00DFBB3C Sending Request msg BYE/cseq=15914 (tdta00E08860) ?17:12:30.747??? tsx00E01944 Transaction created for Request msg BYE/cseq=15913 (tdta00E08860) ?17:12:30.747??? tsx00E01944 Sending Request msg BYE/cseq=15913 (tdta00E08860) in state Null ?17:12:30.747? sip_resolve.c Target 'Ip2:0' type=Unspecified resolved to 'Ip2:5060' type=UDP (UDP transport) ?17:12:30.747?? pjsua_core.c TX 383 bytes Request msg BYE/cseq=15913 (tdta00E08860) to UDP Ip2:5060: BYE sip:anonymous at Ip2 SIP/2.0 Via: SIP/2.0/UDP Ip3:5060;rport;branch=z9hG4bKPjeb7706e927054c668797c77f78f3d523 Max-Forwards: 70 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 To: sip:3001 at Ip2;tag=2802631630558 Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15913 BYE User-Agent: PJSUA v0.8.0-trunk/win32 Content-Length:? 0 --end msg-- ?17:12:30.747??? tsx00E01944 State changed from Null to Calling, event=TX_MSG ?17:12:30.747??? dlg00DFBB3C Transaction tsx00E01944 state changed to Calling ?17:12:30.747?? pjsua_core.c Shutting down... ?17:12:30.965 sip_endpoint.c Processing incoming message: Response msg 200/BYE/cseq=15913 (rdata00DBDC9C) ?17:12:30.965?? pjsua_core.c RX 342 bytes Response msg 200/BYE/cseq=15913 (rdata00DBDC9C) from UDP Ip2:5060: SIP/2.0 200 OK To: "Anonymous"<sip:3001 at Ip2>;tag=2802631630558 From: <sip:Ip3>;tag=5df159bbd00049d2bd2021b735a169c4 Via: SIP/2.0/UDP Ip3:5060;rport=5060;received=Ip1;branch=z9hG4bKPjeb7706e927054c668797c77f78f3d523 Call-ID: 45ba2656ddd840aba5303585d64c5ba8 CSeq: 15913 BYE Content-Length: 0 --end msg-- ?17:12:30.965??? tsx00E01944 Incoming Response msg 200/BYE/cseq=15913 (rdata00DBDC9C) in state Calling ?17:12:30.965??? tsx00E01944 State changed from Calling to Completed, event=RX_MSG ?17:12:30.965??? dlg00DFBB3C Received Response msg 200/BYE/cseq=15913 (rdata00DBDC9C) ?17:12:30.965??? dlg00DFBB3C Transaction tsx00E01944 state changed to Completed ?17:12:30..965??? pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal call clearing)] ?17:12:30.965??? pjsua_app.c Call 0 disconnected, dumping media stats ? [DISCONNCTD] To: sip:3001 at Ip2;tag=2802631630558 ??? Call time: 00h:00m:17s, 1st res in 484 ms, conn in 484ms ??? SRTP status: Not active Crypto-suite: (null) ??? #0 GSM @8KHz, sendrecv, peer=76.235.156.64:49220 ?????? RX pt=3, stat last update: 00h:00m:00.578s ago ????????? total 1pkt 0B (40B +IP hdr) @avg=0bps/18bps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?????? TX pt=3, ptime=20ms, stat last update: never ????????? total 883pkt 29.1KB (64.4KB +IP hdr) @avg 13.1Kbps/29.1Kbps ????????? pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) ??????????????? (msec)??? min???? avg???? max???? last??? dev ????????? loss period:?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????????? jitter???? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ????? RTT msec?????? :?? 0.000?? 0.000?? 0.000?? 0.000?? 0.000 ?17:12:30.965? pjsua_media.c Media session for call 0 is destroyed ?17:12:30.965?? tdta00E07830 Destroying txdata Request msg ACK/cseq=15912 (tdta00E07830) ?17:12:30.965??? dlg00DFBB3C Session count dec to 1 by mod-invite ?17:12:31.747? pjsua_media.c Closing (null) sound playback device and (null) sound capture device ?17:12:31.762????? pasound.c Stopping stream.. ?17:12:31.762????? pasound.c PA message: WinMME StopStream: waiting for background thread. ?17:12:31.856????? pasound.c PA message: WinMME StopStream: waiting for background thread. ?17:12:31.856????? pasound.c Done, status=0 ?17:12:31.856????? pasound.c Closing Conexant HD Audio input: 0 underflow, 0 overflow ?17:12:32.356????????? media Thread stack max usage=7321 by d:\people\siu\projects\voip\pjsip\pjlib\src\pj\os_core_win32.c:1380 ?17:12:32.356????? pasound.c PortAudio sound library shutting down.. ?17:12:32.356????? pasound.c PA message: TerminateHostApis in ?17:12:32.356????? pasound.c PA message: TerminateHostApis out ?17:12:32.356 sip_endpoint.c Destroying endpoing instance.. ?17:12:32.356 sip_endpoint.c Module "mod-pjsua-options" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-pjsua-im" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-pjsua-pres" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-pjsua" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-stateful-util" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-refer" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-presence" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-evsub" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-invite" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-100rel" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-ua" unregistered ?17:12:32.356 sip_transactio Stopping transaction layer module ?17:12:32.356?? tdta00E08860 Destroying txdata Request msg BYE/cseq=15913 (tdta00E08860) ?17:12:32.356??? tsx00E01944 Transaction destroyed! ?17:12:32.356 sip_transactio Transaction layer module destroyed ?17:12:32.356 sip_endpoint.c Module "mod-tsx-layer" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-msg-print" unregistered ?17:12:32.356 sip_endpoint.c Module "mod-pjsua-log" unregistered ?17:12:32.356 sip_transport. Destroying transport manager ?17:12:32.356??? tcplis:5060 SIP TCP listener destroyed ?17:12:32.372 sip_endpoint.c Endpoint 003C6BFC destroyed ?17:12:32.372??? dlg00DFBAD8 Pool is not released by application, releasing now ?17:12:32.372?? pjsua_core.c PJSUA destroyed... == ----- Original Message ---- From: Benny Prijono <bennylp@xxxxxxxxx> To: pjsip list <pjsip at lists.pjsip.org> Sent: Tuesday, June 10, 2008 12:13:26 PM Subject: Re: (1) PJSUA: Problems to talk with a PocketPC, (2) NAT issue On Tue, Jun 10, 2008 at 7:35 PM, Paul Chen <pocketpc_1 at yahoo.com> wrote: > (2) For this NAT issue, there should be a way to let RTP packets going > through it instead of changing the router. As mentioned before, if I use a > softphone on the PC instead of pjsua with the same settings, that softphone > figures out a way to receive the RTP packets. Just a thought. Sure. If other softphones can work then probably your NAT router is not the nasty one so I'm optimistic that we can make pjsip to work with the right settings. With ICE and TURN in the latest pjsip we should even be able to make it work across nasty NATs. But you've left us pretty much in the dark here with your setup, settings, and the log, hence I asked you these a mail ago. Without these I can't help too much (and yes, it works here without port forwarding). Cheers Benny _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip at lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080611/18236494/attachment-0001.html
