Question:
I run pjsip on linux and call pjsip on windows,calling is ok,but see pjsip on linux send any rtp packet,
why?
Thanks a lot.
Info:
### BEGIN LOG - DATE: 100331, TIME: 111201 ###
./pjsua-i686-pc-linux-gnu
Thread name:thr%p, thread id:0
10:01:32.941 os_core_unix.c pjlib 1.5.5 for POSIX initialized
192.168.1.252
10:01:32.943 sip_endpoint.c Creating endpoint instance...
10:01:32.944 pjlib select() I/O Queue created (0x92b3838)
10:01:32.945 sip_endpoint.c Module "mod-msg-print" registered
10:01:32.945 sip_transport. Transport manager created.
10:01:32.946 sip_endpoint.c Module "mod-pjsua-log" registered
10:01:32.947 sip_endpoint.c Module "mod-tsx-layer" registered
10:01:32.947 sip_endpoint.c Module "mod-stateful-util" registered
10:01:32.948 sip_endpoint.c Module "mod-ua" registered
10:01:32.948 sip_endpoint.c Module "mod-100rel" registered
10:01:32.949 sip_endpoint.c Module "mod-pjsua" registered
10:01:32.949 sip_endpoint.c Module "mod-invite" registered
10:01:32.998 pa_dev.c PortAudio sound library initialized, status=0
10:01:32.999 pa_dev.c PortAudio host api count=2
10:01:33.000 pa_dev.c Sound device count=6
10:01:33.001 pjlib select() I/O Queue created (0x92cdd34)
10:01:33.012 sip_endpoint.c Module "mod-evsub" registered
10:01:33.012 sip_endpoint.c Module "mod-presence" registered
10:01:33.013 sip_endpoint.c Module "mod-mwi" registered
10:01:33.014 sip_endpoint.c Module "mod-refer" registered
10:01:33.014 sip_endpoint.c Module "mod-pjsua-pres" registered
10:01:33.015 sip_endpoint.c Module "mod-pjsua-im" registered
10:01:33.016 sip_endpoint.c Module "mod-pjsua-options" registered
10:01:33.016 pjsua_core.c 1 SIP worker threads created
10:01:33.017 pjsua_core.c pjsua version 1.5.5 for i686-pc-linux-gnu initialized
10:01:33.018 sip_endpoint.c Module "mod-default-handler" registered
192.168.1.252
10:01:33.020 pjsua_core.c SIP UDP socket reachable at 192.168.1.252:5060
10:01:33.020 udp0x92dd240 SIP UDP transport started, published address is 192.168.1.252:5060
10:01:33.021 pjsua_acc.c Account <sip:192.168.1.252:5060> added with id 0
192.168.1.252
10:01:33.024 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.1.252:5060
10:01:33.026 pjsua_acc.c Account <sip:192.168.1.252:5060;transport=TCP> added with id 1
192.168.1.252
10:01:33.027 pjsua_media.c RTP socket reachable at 192.168.1.252:4000
10:01:33.028 pjsua_media.c RTCP socket reachable at 192.168.1.252:4001
192.168.1.252
10:01:33.029 pjsua_media.c RTP socket reachable at 192.168.1.252:4002
10:01:33.030 pjsua_media.c RTCP socket reachable at 192.168.1.252:4003
192.168.1.252
10:01:33.032 pjsua_media.c RTP socket reachable at 192.168.1.252:4004
10:01:33.033 pjsua_media.c RTCP socket reachable at 192.168.1.252:4005
192.168.1.252
10:01:33.034 pjsua_media.c RTP socket reachable at 192.168.1.252:4006
10:01:33.035 pjsua_media.c RTCP socket reachable at 192.168.1.252:4007
10:01:33.035 sip_endpoint.c Module "mod-unsolicited-mwi" registered
>>>>
Account list:
[ 0] <sip:192.168.1.252:5060>: does not register
Online status: Online
*[ 1] <sip:192.168.1.252:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 0 active call
>>> m
(You currently have 0 calls)
Buddy list:
-none-
Choices:
0 For current dialog.
-1 All 0 buddies in buddy list
[1 - 0] Select from buddy list
URL An URL
<Enter> Empty input (or 'q') to cancel
Make call: sip:192.168.1.156
10:01:50.855 pjsua_media.c Opening sound device PCM at 16000/2/10ms
Pa_OpenStream returned:
*(PaStream** stream): 0x(nil)
PaError: -9997 ( Invalid sample rate )
Pa_OpenStream returned:
*(PaStream** stream): 0x0x92eaad8
PaError: 0 ( Success )
Pa_OpenStream returned:
*(PaStream** stream): 0x(nil)
PaError: -9997 ( Invalid sample rate )
10:01:50.866 pjsua_media.c Opening sound device PCM at 44100/2/10ms
Pa_OpenStream returned:
*(PaStream** stream): 0x(nil)
PaError: -9997 ( Invalid sample rate )
Pa_OpenStream returned:
*(PaStream** stream): 0x(nil)
PaError: -9997 ( Invalid sample rate )
10:01:50.871 pjsua_media.c Opening sound device PCM at 48000/2/10ms
Pa_OpenStream returned:
*(PaStream** stream): 0x0x92eb2d8
PaError: 0 ( Success )
10:01:50.879 echo_speex.c Multichannel EC is not supported by this echo canceller. It may not work.
10:01:50.889 ec0x92cd758 AEC created, clock_rate=48000, channel=2, samples per frame=960, tail length=200 ms, latency=140 ms
StartStream:1
10:01:50.894 pjsua_call.c Making call with acc #1 to sip:192.168.1.156
10:01:50.895 pjsua_media.c Media index 0 selected for call 0
10:01:50.897 pjsua_core.c TX 1073 bytes Request msg INVITE/cseq=20430 (tdta0x937d958) to UDP 192.168.1.156:5060:
INVITE sip:192.168.1.156 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;rport;branch=z9hG4bKPjda0dd5ed-6312-43db-94b9-5f0bd4a9243a
Max-Forwards: 70
From: <sip:192.168.1.252>;tag=36869b20-9582-496d-af7b-16556ddb0398
To: sip:192.168.1.156
Contact: <sip:192.168.1.252>
Call-ID: 4a071e22-35fc-426d-9aca-8802778b6678
CSeq: 20430 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.5.5/i686-pc-linux-gnu
Content-Type: application/sdp
Content-Length: 462
v=0
o=- 3478989710 3478989710 IN IP4 192.168.1.252
s=pjmedia
c=IN IP4 192.168.1.252
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 192.168.1.252
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--
10:01:50.898 pjsua_app.c Call 0 state changed to CALLING
>>> FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
Thread name:pa_rec, thread id:0
rec_num: 1
put frame num: 1
Thread name:portaudio, thread id:0
10:01:50.944 os_core_unix.c Info: possibly re-registering existing thread
play_num: 1
put frame num: 1
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3326 num:1
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
put frame num: 11
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
10:01:51.071 pjsua_core.c RX 326 bytes Response msg 100/INVITE/cseq=20430 (rdata0x92dd6b4) from UDP 192.168.1.156:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.252:5060;rport=5060;received=192.168.1.252;branch=z9hG4bKPjda0dd5ed-6312-43db-94b9-5f0bd4a9243a
Call-ID: 4a071e22-35fc-426d-9aca-8802778b6678
From: <sip:192.168.1.252>;tag=36869b20-9582-496d-af7b-16556ddb0398
To: <sip:192.168.1.156>
CSeq: 20430 INVITE
Content-Length: 0
--end msg--
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3343
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3407
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3416
FILE:src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c line:3292
10:02:02.031 udp0x92dee78 Remote RTCP address switched to 192.168.1.156:4001
10:02:02.056 pjsua_core.c RX 859 bytes Response msg 200/INVITE/cseq=20430 (rdata0x92dd6b4) from UDP 192.168.1.156:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;rport=5060;received=192.168.1.252;branch=z9hG4bKPjda0dd5ed-6312-43db-94b9-5f0bd4a9243a
Call-ID: 4a071e22-35fc-426d-9aca-8802778b6678
From: <sip:192.168.1.252>;tag=36869b20-9582-496d-af7b-16556ddb0398
To: <sip:192.168.1.156>;tag=7d88b07583bb4e1a96c053d641107864
CSeq: 20430 INVITE
Contact: <sip:192.168.1.156:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 254
v=0
o=- 3479022756 3479022757 IN IP4 192.168.1.156
s=pjmedia
c=IN IP4 192.168.1.156
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
a=rtcp:4001 IN IP4 192.168.1.156
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--end msg--
10:02:02.056 pjsua_app.c Call 0 state changed to CONNECTING
10:02:02.058 strm0x9381d34 VAD temporarily disabled
10:02:02.059 strm0x9381d34 Encoder stream started
10:02:02.060 strm0x9381d34 Decoder stream started
10:02:02.060 pjsua_media.c Media updates, stream #0: PCMU (sendrecv)
10:02:02.060 conference.c Port 3 (sip:192.168.1.156) transmitting to port 0 (Ensoniq AudioPCI: ES1371 DAC2/ADC (hw:0,0) (48KHz))
10:02:02.060 conference.c Port 0 (Ensoniq AudioPCI: ES1371 DAC2/ADC (hw:0,0) (48KHz)) transmitting to port 3 (sip:192.168.1.156)
10:02:02.060 pjsua_app.c Media for call 0 is active
10:02:02.060 pjsua_core.c TX 364 bytes Request msg ACK/cseq=20430 (tdta0x93858f8) to UDP 192.168.1.156:5060:
ACK sip:192.168.1.156:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;rport;branch=z9hG4bKPj2e3bced0-36e8-4754-957f-0539274ad5b7
Max-Forwards: 70
From: <sip:192.168.1.252>;tag=36869b20-9582-496d-af7b-16556ddb0398
To: sip:192.168.1.156;tag=7d88b07583bb4e1a96c053d641107864
Call-ID: 4a071e22-35fc-426d-9aca-8802778b6678
CSeq: 20430 ACK
Content-Length: 0
--end msg--
10:02:02.061 pjsua_app.c Call 0 state changed to CONFIRMED
### END LOG - DATE: 100331, TIME: 111307 ###
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