Hello,
I am using the pjsua application to connect another machine in a LAN for a
peer-to-peer call. I have checked all the playback and recording
functionalities locally at both ends and all are perfect.
When I call another machine with IP address, the call gets connected and
prompt is received at the rcvr end for answering the call. Then it asks to
send a code (100-699) and after sending no sound is heard either ways.
I would like to know what does this code (100-699) imply!
Also, in the whole process only beep sound is heard continuously.
I am printing the console messages at both ends.
At call initiator end:
******************************************************************************
[user at localhost bin]$ ./pjsua-i686-pc-linux-gnu sip:192.168.0.16
15:43:40.044 os_core_unix.c pjlib 1.0.3 for POSIX initialized
15:43:40.045 sip_endpoint.c Creating endpoint instance...
15:43:40.045 pjlib select() I/O Queue created (0x915b1d0)
15:43:40.045 sip_endpoint.c Module "mod-msg-print" registered
15:43:40.045 sip_transport. Transport manager created.
15:43:40.045 sip_endpoint.c Module "mod-pjsua-log" registered
15:43:40.045 sip_endpoint.c Module "mod-tsx-layer" registered
15:43:40.045 sip_endpoint.c Module "mod-stateful-util" registered
15:43:40.045 sip_endpoint.c Module "mod-ua" registered
15:43:40.045 sip_endpoint.c Module "mod-100rel" registered
15:43:40.045 sip_endpoint.c Module "mod-pjsua" registered
15:43:40.045 sip_endpoint.c Module "mod-invite" registered
15:43:40.084 pasound.c PortAudio sound library initialized, status=0
15:43:40.084 pasound.c PortAudio host api count=2
15:43:40.084 pasound.c Sound device count=10
15:43:40.084 pjlib select() I/O Queue created (0x917f974)
15:43:40.084 sip_endpoint.c Module "mod-evsub" registered
15:43:40.084 sip_endpoint.c Module "mod-presence" registered
15:43:40.084 sip_endpoint.c Module "mod-refer" registered
15:43:40.084 sip_endpoint.c Module "mod-pjsua-pres" registered
15:43:40.084 sip_endpoint.c Module "mod-pjsua-im" registered
15:43:40.084 sip_endpoint.c Module "mod-pjsua-options" registered
15:43:40.084 pjsua_core.c 1 SIP worker threads created
15:43:40.084 pjsua_core.c pjsua version 1.0.3 for i686-pc-linux-gnu
initialized
15:43:40.084 sip_endpoint.c Module "mod-default-handler" registered
15:43:40.085 pjsua_core.c SIP UDP socket reachable at 192.168.0.8:5060
15:43:40.085 udp0x9190020 SIP UDP transport started, published address
is 192.168.0.8:5060
15:43:40.085 pjsua_acc.c Account <sip:192.168.0.8:5060> added with id 0
15:43:40.085 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.0.8:5060
15:43:40.085 pjsua_acc.c Account <sip:192.168.0.8:5060;transport=TCP>
added with id 1
15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4000
15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4001
15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4002
15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4003
15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4004
15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4005
15:43:40.085 pjsua_media.c RTP socket reachable at 192.168.0.8:4006
15:43:40.085 pjsua_media.c RTCP socket reachable at 192.168.0.8:4007
15:43:40.085 pjsua_media.c pjsua_set_snd_dev(): attempting to open
devices @16000 Hz
15:43:40.088 pjsua_media.c ..failed: Invalid sample rate
15:43:40.088 pjsua_media.c pjsua_set_snd_dev(): attempting to open
devices @44100 Hz
15:43:40.128 os_core_unix.c Info: possibly re-registering existing thread
15:43:40.217 ec0x917ee18 AEC created, clock_rate=44100, channel=1,
samples per frame=882, tail length=200 ms, latency=88969 ms
15:43:40.217 pjsua_call.c Making call with acc #1 to sip:192.168.0.16
15:43:40.228 pjsua_media.c Media index 0 selected for call 0
15:43:40.228 pjsua_core.c TX 1020 bytes Request msg INVITE/cseq=431
(tdta0x9ad4d40) to UDP 192.168.0.16:5060:
INVITE sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16
Contact: <sip:192.168.0.8:5060>
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3497111020 3497111020 IN IP4 192.168.0.8
s=pjmedia
c=IN IP4 192.168.0.8
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 192.168.0.8
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--
15:43:40.228 pjsua_app.c Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.0.8:5060>: does not register
Online status: Online
*[ 1] <sip:192.168.0.8:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:192.168.0.16
+=============================================================================+
| 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 sleep MS echo [0|1|txt] n: detect NAT type
|
+=============================================================================+
You have 1 active call
Current call id=0 to sip:192.168.0.16 [CALLING]
>>> 15:43:40.241 pjsua_core.c RX 317 bytes Response msg
100/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:43:45.229 sound_port.c EC suspended because of inactivity
15:43:51.065 pjsua_core.c RX 317 bytes Response msg
100/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:44:11.801 pjsua_core.c RX 359 bytes Response msg
603/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:44:11.801 pjsua_core.c TX 355 bytes Request msg ACK/cseq=431
(tdta0x9ad74f8) to UDP 192.168.0.16:5060:
ACK sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 ACK
Content-Length: 0
--end msg--
15:44:11.801 pjsua_app.c Call 0 is DISCONNECTED [reason=603 (Decline)]
15:44:11.801 pjsua_app.c
[DISCONNCTD] To: sip:192.168.0.16
Call time: 00h:00m:00s, 1st res in 31584 ms, conn in 0ms
SRTP status: Not active Crypto-suite: (null)
15:44:13.305 pjsua_core.c RX 359 bytes Response msg
603/INVITE/cseq=431 (rdata0x9190494) from UDP 192.168.0.16:5060:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:44:13.305 pjsua_core.c TX 355 bytes Request msg ACK/cseq=431
(tdta0x9ad74f8) to UDP 192.168.0.16:5060:
ACK sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 ACK
Content-Length: 0
--end msg--
q
15:44:18.332 pjsua_media.c Closing (null) sound playback device and
(null) sound capture device
15:44:19.638 pasound.c PortAudio sound library shutting down..
15:44:19.638 pjsua_core.c Shutting down...
15:44:20.645 pjsua_core.c Destroying...
15:44:20.645 sip_transactio Stopping transaction layer module
15:44:20.646 sip_endpoint.c Module "mod-default-handler" unregistered
15:44:20.646 sip_endpoint.c Module "mod-pjsua-options" unregistered
15:44:20.646 sip_endpoint.c Module "mod-pjsua-im" unregistered
15:44:20.646 sip_endpoint.c Module "mod-pjsua-pres" unregistered
15:44:20.646 sip_endpoint.c Module "mod-pjsua" unregistered
15:44:20.646 sip_endpoint.c Module "mod-stateful-util" unregistered
15:44:20.646 sip_endpoint.c Module "mod-refer" unregistered
15:44:20.646 sip_endpoint.c Module "mod-presence" unregistered
15:44:20.646 sip_endpoint.c Module "mod-evsub" unregistered
15:44:20.646 sip_endpoint.c Module "mod-invite" unregistered
15:44:20.646 sip_endpoint.c Module "mod-100rel" unregistered
15:44:20.646 sip_endpoint.c Module "mod-ua" unregistered
15:44:20.646 sip_transactio Transaction layer module destroyed
15:44:20.646 sip_endpoint.c Module "mod-tsx-layer" unregistered
15:44:20.646 sip_endpoint.c Module "mod-msg-print" unregistered
15:44:20.646 sip_endpoint.c Module "mod-pjsua-log" unregistered
15:44:20.647 tcplis:5060 SIP TCP listener destroyed
15:44:20.647 sip_endpoint.c Endpoint 0x9153324 destroyed
15:44:20.647 pjsua_core.c PJSUA destroyed...
[user at localhost bin]$
***************************************************************************
At the call receiver end:
****************************************************************************
[user at localhost bin]$ ./pjsua-i686-pc-linux-gnu
15:48:56.123 os_core_unix.c pjlib 1.0.3 for POSIX initialized
15:48:56.123 sip_endpoint.c Creating endpoint instance...
15:48:56.124 pjlib select() I/O Queue created (0x86fd1d0)
15:48:56.124 sip_endpoint.c Module "mod-msg-print" registered
15:48:56.124 sip_transport. Transport manager created.
15:48:56.124 sip_endpoint.c Module "mod-pjsua-log" registered
15:48:56.124 sip_endpoint.c Module "mod-tsx-layer" registered
15:48:56.124 sip_endpoint.c Module "mod-stateful-util" registered
15:48:56.124 sip_endpoint.c Module "mod-ua" registered
15:48:56.124 sip_endpoint.c Module "mod-100rel" registered
15:48:56.124 sip_endpoint.c Module "mod-pjsua" registered
15:48:56.124 sip_endpoint.c Module "mod-invite" registered
15:48:56.164 pasound.c PortAudio sound library initialized, status=0
15:48:56.164 pasound.c PortAudio host api count=2
15:48:56.164 pasound.c Sound device count=10
15:48:56.164 pjlib select() I/O Queue created (0x872192c)
15:48:56.164 sip_endpoint.c Module "mod-evsub" registered
15:48:56.164 sip_endpoint.c Module "mod-presence" registered
15:48:56.164 sip_endpoint.c Module "mod-refer" registered
15:48:56.164 sip_endpoint.c Module "mod-pjsua-pres" registered
15:48:56.164 sip_endpoint.c Module "mod-pjsua-im" registered
15:48:56.164 sip_endpoint.c Module "mod-pjsua-options" registered
15:48:56.164 pjsua_core.c 1 SIP worker threads created
15:48:56.164 pjsua_core.c pjsua version 1.0.3 for i686-pc-linux-gnu
initialized
15:48:56.164 sip_endpoint.c Module "mod-default-handler" registered
15:48:56.164 pjsua_core.c SIP UDP socket reachable at 192.168.0.16:5060
15:48:56.164 udp0x87320d0 SIP UDP transport started, published address
is 192.168.0.16:5060
15:48:56.165 pjsua_acc.c Account <sip:192.168.0.16:5060> added with id 0
15:48:56.165 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.0.16:5060
15:48:56.165 pjsua_acc.c Account
<sip:192.168.0.16:5060;transport=TCP> added with id 1
15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4000
15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4001
15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4002
15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4003
15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4004
15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4005
15:48:56.165 pjsua_media.c RTP socket reachable at 192.168.0.16:4006
15:48:56.165 pjsua_media.c RTCP socket reachable at 192.168.0.16:4007
>>>>
Account list:
[ 0] <sip:192.168.0.16:5060>: does not register
Online status: Online
*[ 1] <sip:192.168.0.16:5060;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 sleep MS echo [0|1|txt] n: detect NAT type
|
+=============================================================================+
You have 0 active call
>>> 15:49:29.163 pjsua_core.c RX 1020 bytes Request msg
INVITE/cseq=431 (rdata0x8732544) from UDP 192.168.0.8:5060:
INVITE sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16
Contact: <sip:192.168.0.8:5060>
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3497111020 3497111020 IN IP4 192.168.0.8
s=pjmedia
c=IN IP4 192.168.0.8
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 192.168.0.8
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--
15:49:29.173 pjsua_media.c Media index 0 selected for call 0
15:49:29.173 pjsua_core.c TX 317 bytes Response msg
100/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:49:29.173 pjsua_media.c pjsua_set_snd_dev(): attempting to open
devices @16000 Hz
15:49:29.176 pjsua_media.c ..failed: Invalid sample rate
15:49:29.176 pjsua_media.c pjsua_set_snd_dev(): attempting to open
devices @44100 Hz
15:49:29.208 os_core_unix.c Info: possibly re-registering existing thread
15:49:29.296 ec0x8720d98 AEC created, clock_rate=44100, channel=1,
samples per frame=882, tail length=200 ms, latency=88969 ms
15:49:29.296 conference.c Port 2 (ring) transmitting to port 0 (HDA
Intel: AD198x Analog (hw:0,0) (44KHz))
15:49:29.296 pjsua_app.c Incoming call for account 0!
From: <sip:192.168.0.8>
To: <sip:192.168.0.16>
Press a to answer or h to reject call
a
Answer with code (100-699) (empty to cancel): 100
15:49:39.999 pjsua_core.c TX 317 bytes Response msg
100/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>
CSeq: 431 INVITE
Content-Length: 0
--end msg--
>>> q
15:50:00.736 pjsua_core.c TX 359 bytes Response msg
603/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:50:00.736 pjsua_app.c Call 0 is DISCONNECTED [reason=603 (Decline)]
15:50:00.736 pjsua_app.c
[DISCONNCTD] To: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
Call time: 00h:00m:00s, 1st res in 10836 ms, conn in 0ms
SRTP status: Not active Crypto-suite: (null)
15:50:00.736 pjsua_media.c Closing (null) sound playback device and
(null) sound capture device
15:50:02.239 pasound.c PortAudio sound library shutting down..
15:50:02.240 pjsua_core.c Shutting down...
15:50:02.240 pjsua_core.c TX 359 bytes Response msg
603/INVITE/cseq=431 (tdta0x8744a48) to UDP 192.168.0.8:5060:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
192.168.0.8:5060;rport=5060;received=192.168.0.8;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: <sip:192.168.0.16>;tag=f768566f-8453-421f-a2d2-f8776a31ff19
CSeq: 431 INVITE
Content-Length: 0
--end msg--
15:50:02.240 pjsua_core.c RX 355 bytes Request msg ACK/cseq=431
(rdata0x8732544) from UDP 192.168.0.8:5060:
ACK sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 ACK
Content-Length: 0
--end msg--
15:50:02.240 pjsua_core.c RX 355 bytes Request msg ACK/cseq=431
(rdata0x8732544) from UDP 192.168.0.8:5060:
ACK sip:192.168.0.16 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.8:5060;rport;branch=z9hG4bKPj35b25afd-d044-4430-9f15-d729cc67d6d6
Max-Forwards: 70
From: <sip:192.168.0.8>;tag=7748b90b-3706-46fa-ae05-48c77190e146
To: sip:192.168.0.16;tag=f768566f-8453-421f-a2d2-f8776a31ff19
Call-ID: 8dc02fb6-f8e0-4a9c-8e1e-d37f36b1f462
CSeq: 431 ACK
Content-Length: 0
--end msg--
15:50:03.248 pjsua_core.c Destroying...
15:50:03.248 sip_transactio Stopping transaction layer module
15:50:03.248 sip_endpoint.c Module "mod-default-handler" unregistered
15:50:03.248 sip_endpoint.c Module "mod-pjsua-options" unregistered
15:50:03.248 sip_endpoint.c Module "mod-pjsua-im" unregistered
15:50:03.248 sip_endpoint.c Module "mod-pjsua-pres" unregistered
15:50:03.248 sip_endpoint.c Module "mod-pjsua" unregistered
15:50:03.248 sip_endpoint.c Module "mod-stateful-util" unregistered
15:50:03.248 sip_endpoint.c Module "mod-refer" unregistered
15:50:03.248 sip_endpoint.c Module "mod-presence" unregistered
15:50:03.248 sip_endpoint.c Module "mod-evsub" unregistered
15:50:03.248 sip_endpoint.c Module "mod-invite" unregistered
15:50:03.248 sip_endpoint.c Module "mod-100rel" unregistered
15:50:03.248 sip_endpoint.c Module "mod-ua" unregistered
15:50:03.248 sip_transactio Transaction layer module destroyed
15:50:03.248 sip_endpoint.c Module "mod-tsx-layer" unregistered
15:50:03.248 sip_endpoint.c Module "mod-msg-print" unregistered
15:50:03.248 sip_endpoint.c Module "mod-pjsua-log" unregistered
15:50:03.249 tcplis:5060 SIP TCP listener destroyed
15:50:03.249 sip_endpoint.c Endpoint 0x86f5324 destroyed
15:50:03.249 pjsua_core.c PJSUA destroyed...
[user at localhost bin]$
****************************************************************************
Any help will be highly appreciated!
Thanks and Regards,
Abhishek Bhattacharya
