PJSIP LOG after executed command vid call add
>>>>
Account list:
*[ 0] sip:6002 at 192.168.2.134:5060: 200/OK (expires=275)
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 | |
+-----------------------------------------------------------------------------+
| Video: "vid help" for more info |
+-----------------------------------------------------------------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 1 active call
Current call id=0 to sip:6005 at 192.168.2.134:5060 [CONFIRMED]
>>> vid call add
18:55:55.533 pjsua_vid.c !Call 0: set video stream, op=1
18:55:55.542 pjsua_media.c .RTP socket reachable at 192.168.2.122:4004
18:55:55.542 pjsua_media.c .RTCP socket reachable at 192.168.2.122:4005
18:55:55.543 pjsua_core.c ....TX 1078 bytes Request msg INVITE/cseq=8630 (tdt
a028B8480) to UDP 192.168.2.134:5060:
INVITE sip:6005 at 192.168.2.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.122:5060;rport;branch=z9hG4bKPja510928b36a744af84e31c
6371ebd0d1
Max-Forwards: 70
From: sip:6002@192.168.2.134;tag=1e2045f27bc14c5392550e5d21270200
To: sip:6005 at 192.168.2.134;tag=as76ce4ed9
Contact: <sip:6002 at 192.168.2.122:5060;ob>
Call-ID: 64e1047404bd45f89da32a5e0d32a6ca
CSeq: 8630 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAG
E, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 469
v=0
o=- 3534691964 3534691965 IN IP4 192.168.2.122
s=pjmedia
c=IN IP4 192.168.2.122
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 96
c=IN IP4 192.168.2.122
a=rtcp:4001 IN IP4 192.168.2.122
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
m=video 0 RTP/AVP 96
c=IN IP4 192.168.2.122
m=video 4004 RTP/AVP 96
c=IN IP4 192.168.2.122
a=rtcp:4005 IN IP4 192.168.2.122
a=sendrecv
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF=1;QCIF=1
--end msg--
>>> 18:55:55.549 pjsua_core.c .RX 545 bytes Response msg 100/INVITE/cseq=8630
(rdata028B54D4) from UDP 192.168.2.134:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.122:5060;branch=z9hG4bKPja510928b36a744af84e31c6371eb
d0d1;received=192.168.2.122;rport=5060
From: sip:6002@192.168.2.134;tag=1e2045f27bc14c5392550e5d21270200
To: sip:6005 at 192.168.2.134;tag=as76ce4ed9
Call-ID: 64e1047404bd45f89da32a5e0d32a6ca
CSeq: 8630 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6005 at 192.168.2.134>
Content-Length: 0
--end msg--
18:55:55.551 pjsua_core.c .RX 869 bytes Response msg 200/INVITE/cseq=8630 (rd
ata028B54D4) from UDP 192.168.2.134:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.122:5060;branch=z9hG4bKPja510928b36a744af84e31c6371eb
d0d1;received=192.168.2.122;rport=5060
From: sip:6002@192.168.2.134;tag=1e2045f27bc14c5392550e5d21270200
To: sip:6005 at 192.168.2.134;tag=as76ce4ed9
Call-ID: 64e1047404bd45f89da32a5e0d32a6ca
CSeq: 8630 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6005 at 192.168.2.134>
Content-Type: application/sdp
Content-Length: 273
v=0
o=root 1746503561 1746503562 IN IP4 192.168.2.134
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 192.168.2.134
t=0 0
m=audio 11470 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 96
--end msg--
18:55:55.580 pjsua_media.c .....Call 0: updating media..
18:55:55.580 pjsua_media.c .......Media session call00:0 is destroyed
18:55:55.581 pjsua_media.c .......Media session call00:1 is destroyed
18:55:55.581 pjsua_media.c .......Media session call00:2 is destroyed
18:55:55.582 pjsua_media.c ......Audio channel update..
18:55:55.582 strm0290EEAC .......VAD temporarily disabled
18:55:55.583 strm0290EEAC .......Encoder stream started
18:55:55.612 strm0290EEAC .......Decoder stream started
18:55:55.613 pjsua_media.c .......Audio updated, stream #0: PCMU (sendrecv)
18:55:55.613 pjsua_vid.c ......Video channel update..
18:55:55.614 pjsua_vid.c .......Video updated, stream #1: (inactive)
18:55:55.614 pjsua_vid.c ......Video channel update..
18:55:55.614 pjsua_vid.c .......Video updated, stream #2: (inactive)
18:55:55.615 pjsua_app.c .....Call 0 media 0 [type=audio], status is Active
18:55:55.616 pjsua_media.c .....Conf connect: 3 --> 0
18:55:55.616 conference.c ......Port 3 (sip:6005 at 192.168.2.134:5060) transmit
ting to port 0 (Wave mapper)
18:55:55.616 pjsua_media.c .....Conf connect: 0 --> 3
18:55:55.617 conference.c ......Port 0 (Wave mapper) transmitting to port 3 (
sip:6005 at 192.168.2.134:5060)
18:55:55.617 pjsua_app.c .....Call 0 media 1 [type=video], status is None
18:55:55.618 pjsua_app.c .....Call 0 media 2 [type=video], status is None
18:55:55.646 pjsua_core.c .....TX 337 bytes Request msg ACK/cseq=8630 (tdta02
8FCBE8) to UDP 192.168.2.134:5060:
ACK sip:6005 at 192.168.2.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.122:5060;rport;branch=z9hG4bKPjc18dde25ce224bd7b30864
28265d3685
Max-Forwards: 70
From: sip:6002@192.168.2.134;tag=1e2045f27bc14c5392550e5d21270200
To: sip:6005 at 192.168.2.134;tag=as76ce4ed9
Call-ID: 64e1047404bd45f89da32a5e0d32a6ca
CSeq: 8630 ACK
Content-Length: 0
