Hello,
In continuation with the below mail regarding my problem of no sound heard
at both ends, I have the following issues:
(1) I have connected both the machines in a LAN environment without any
firewalls and able to ping each other. My call initiator machine is
192.168.0.8 and call recvr machine is 192.168.0.16.
I am performing: ./pjsua sip:192.168.0.16 from call initiator machine. Is
this correct?
(2) I have gone thorough the documentation for solving audio problems and
tested all of them. Everything is fine as regards to local record/playback
at both the machines. But after initiating ./pjsua sip:192.168.0.16 the
ports at the initiator machine with cl command look as follows:
Conference ports:
Port ##00[16KHz/20ms/1] HDA Intel: AD198X Analog (hw:0,0) (44KHz)
transmitting to:
Port ##01[16KHz/20ms/1] ringback transmitting to:
Port ##02[16KHz/20ms/1] ring transmitting to:
But according to the documentation, there should be one entry in the
conference bridge with the destination sip connection. So, I am also not
able to perform cc for connecting the call to the devices. Why this
problem is happening?
(3) Even though, a prompt appears at the rcvr end for accepting the call
and after pressing a to accept then it asks for code (100-699) and on
entering 100 nothing happens and a beep sound continues at the rcvr end.
What does this code (100-699) imply and how should I connect? By entering
100 should I be able to connect?
Even though the call prompt is received means that network connection is
OK, then why the call is not connected to the conference bridge and no
sound is heard?
Your help is highly appreciated,
Thanks,
Abhishek
---------------------------- Original Message ----------------------------
Subject: No sound is heard at local/remote end!
From: "Abhishek Bhattacharya" <abhat002@xxxxxxxxxxx>
Date: Tue, October 26, 2010 4:32 pm
To: pjsip at lists.pjsip.org
--------------------------------------------------------------------------
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
Regards,
Abhishek Bhattacharya
