Hi,
I am using PJSUA on Linux machine for calling a X-lite subscriber. I intend
to play a wav file on the sip-phone as it answers the call or anytime during
the call; like we have in IVRS applications. For this I have tried using
pjsua with the following options:
./pjsua --id=sip:user1 at example.com
<sip%3Auser1 at example.com>--play-file=/home/cdot/sample.wav
where example.com is defined as the domain name of my system(in /etc/hosts)
and sample.wav is a wav file that I need to play.
next when i place a call using 'm' and to sip url of x-lite, lets say
"sip:4 at titikshu"
*The call is getting placed but I am not able to hear any wav file on the
x-lite phone*.
*Could you explain the reason for the same and what changes I would need in
order to resolve t5he problem and achieve the desired results.
*
The logs displayed on my system are as follows:
./pjsua-x86_64-unknown-linux-gnu
--id=sip:user1 at example.com<sip%3Auser1 at example.com>--play-file=/home/cdot/sample.wav
16:46:13.021 os_core_unix.c pjlib 1.6 for POSIX initialized
16:46:13.035 sip_endpoint.c Creating endpoint instance...
16:46:13.036 pjlib select() I/O Queue created (0x18e3c710)
16:46:13.036 sip_endpoint.c Module "mod-msg-print" registered
16:46:13.036 sip_transport. Transport manager created.
16:46:13.036 sip_endpoint.c Module "mod-pjsua-log" registered
16:46:13.036 sip_endpoint.c Module "mod-tsx-layer" registered
16:46:13.036 sip_endpoint.c Module "mod-stateful-util" registered
16:46:13.036 sip_endpoint.c Module "mod-ua" registered
16:46:13.036 sip_endpoint.c Module "mod-100rel" registered
16:46:13.036 sip_endpoint.c Module "mod-pjsua" registered
16:46:13.036 sip_endpoint.c Module "mod-invite" registered
16:46:13.103 pa_dev.c PortAudio sound library initialized, status=0
16:46:13.103 pa_dev.c PortAudio host api count=2
16:46:13.103 pa_dev.c Sound device count=11
16:46:13.103 pjlib select() I/O Queue created (0x18e775d8)
16:46:13.117 sip_endpoint.c Module "mod-evsub" registered
16:46:13.117 sip_endpoint.c Module "mod-presence" registered
16:46:13.117 sip_endpoint.c Module "mod-mwi" registered
16:46:13.117 sip_endpoint.c Module "mod-refer" registered
16:46:13.117 sip_endpoint.c Module "mod-pjsua-pres" registered
16:46:13.117 sip_endpoint.c Module "mod-pjsua-im" registered
16:46:13.117 sip_endpoint.c Module "mod-pjsua-options" registered
16:46:13.117 pjsua_core.c 1 SIP worker threads created
16:46:13.117 pjsua_core.c pjsua version 1.6 for x86_64-unknown-linux-gnu
initialized
16:46:13.117 sip_endpoint.c Module "mod-default-handler" registered
16:46:13.117 wav_player.c File player '/home/cdot/sample.wav' created:
samp.rate=22050, ch=1, bufsize=4KB, filesize=121KB
16:46:13.118 pjsua_core.c SIP UDP socket reachable at
192.168.105.90:5060
16:46:13.118 udp0x18e8d040 SIP UDP transport started, published address
is 192.168.105.90:5060
16:46:13.118 pjsua_acc.c Account <sip:192.168.105.90:5060> added with
id 0
16:46:13.118 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.105.90:5060
16:46:13.118 pjsua_acc.c Account <sip:192.168.105.90:5060;transport=TCP>
added with id 1
16:46:13.118 pjsua_acc.c Account
sip:user1 at example.com<sip%3Auser1 at example.com>added with id 2
16:46:13.118 pjsua_media.c RTP socket reachable at 192.168.105.90:4000
16:46:13.118 pjsua_media.c RTCP socket reachable at 192.168.105.90:4001
16:46:13.118 pjsua_media.c RTP socket reachable at 192.168.105.90:4002
16:46:13.118 pjsua_media.c RTCP socket reachable at 192.168.105.90:4003
16:46:13.118 pjsua_media.c RTP socket reachable at 192.168.105.90:4004
16:46:13.118 pjsua_media.c RTCP socket reachable at 192.168.105.90:4005
16:46:13.118 pjsua_media.c RTP socket reachable at 192.168.105.90:4006
16:46:13.118 pjsua_media.c RTCP socket reachable at 192.168.105.90:4007
16:46:13.118 sip_endpoint.c Module "mod-unsolicited-mwi" registered
>>>>
Account list:
[ 0] <sip:192.168.105.90:5060>: does not register
Online status: Online
[ 1] <sip:192.168.105.90:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:user1 at example.com <sip%3Auser1 at example.com>: 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:4 at titikshu
16:46:24.564 pjsua_media.c Opening sound device PCM at 16000/1/20ms
16:46:24.571 ec0x18e76cd0 AEC created, clock_rate=16000, channel=1,
samples per frame=320, tail length=200 ms, latency=100 ms
16:46:24.576 pjsua_call.c Making call with acc #2 to sip:4 at titikshu
16:46:24.576 pjsua_media.c Media index 0 selected for call 0
16:46:24.577 pjsua_core.c TX 1090 bytes Request msg INVITE/cseq=26487
(tdta0x18ecf590) to UDP 196.1.110.214:5060:
INVITE sip:4 at titikshu SIP/2.0
Via: SIP/2.0/UDP 192.168.105.90:5060
;rport;branch=z9hG4bKPjd5989ccd-d3ec-47b4-9107-1f793bf9f9b5
Max-Forwards: 70
From: sip:user1@xxxxxxxxxxx <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
To: sip:4 at titikshu
Contact: <sip:user1 at 192.168.105.90:5060>
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 26487 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 v1.6/x86_64-unknown-linux-gnu
Content-Type: application/sdp
Content-Length: 465
v=0
o=- 3484379784 3484379784 IN IP4 192.168.105.90
s=pjmedia
c=IN IP4 192.168.105.90
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4 192.168.105.90
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 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--
16:46:24.577 pjsua_app.c Call 0 state changed to CALLING
>>> 16:46:24.580 pjsua_core.c RX 379 bytes Response msg
100/INVITE/cseq=26487 (rdata0x18e8d508) from UDP 196.1.110.214:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.105.90:5060
;rport;branch=z9hG4bKPjd5989ccd-d3ec-47b4-9107-1f793bf9f9b5
From: sip:user1@xxxxxxxxxxx <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
To: <sip:4 at titikshu>;tag=1673815833
Contact: <sip:4 at 196.1.110.214:5060>
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 26487 INVITE
Server: X-Lite release 1105d
Content-Length: 0
--end msg--
16:46:24.580 pjsua_app.c Call 0 state changed to EARLY (100 Trying)
16:46:24.612 pjsua_core.c RX 380 bytes Response msg
180/INVITE/cseq=26487 (rdata0x18e8d508) from UDP 196.1.110.214:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.105.90:5060
;rport;branch=z9hG4bKPjd5989ccd-d3ec-47b4-9107-1f793bf9f9b5
From: sip:user1@xxxxxxxxxxx <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
To: <sip:4 at titikshu>;tag=1673815833
Contact: <sip:4 at 196.1.110.214:5060>
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 26487 INVITE
Server: X-Lite release 1105d
Content-Length: 0
--end msg--
16:46:24.612 conference.c Port 2 (ringback) transmitting to port 0 (HDA
Intel: AD198x Analog (hw:0,0))
16:46:24.612 pjsua_app.c Call 0 state changed to EARLY (180 Ringing)
16:46:24.616 os_core_unix.c Info: possibly re-registering existing thread
16:46:29.617 sound_port.c EC suspended because of inactivity
16:46:31.606 udp0x18e8f0d0 Remote RTCP address switched to
196.1.110.214:4001
16:46:31.611 pjsua_core.c RX 689 bytes Response msg
200/INVITE/cseq=26487 (rdata0x18e8d508) from UDP 196.1.110.214:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.105.90:5060
;rport;branch=z9hG4bKPjd5989ccd-d3ec-47b4-9107-1f793bf9f9b5
From: sip:user1@xxxxxxxxxxx <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
To: <sip:4 at titikshu>;tag=1673815833
Contact: <sip:4 at 196.1.110.214:5060>
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 26487 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105d
Content-Length: 281
v=0
o=4 4080949630 4080956661 IN IP4 196.1.110.214
s=X-Lite
c=IN IP4 196.1.110.214
t=0 0
m=audio 4000 RTP/AVP 98 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--end msg--
16:46:31.611 pjsua_app.c Call 0 state changed to CONNECTING
16:46:31.611 strm0x18ed5458 VAD temporarily disabled
16:46:31.611 strm0x18ed5458 Encoder stream started
16:46:31.611 strm0x18ed5458 Decoder stream started
16:46:31.611 pjsua_media.c Media updates, stream #0: iLBC (sendrecv)
16:46:31.611 conference.c Port 2 (ringback) stop transmitting to port 0
(HDA Intel: AD198x Analog (hw:0,0))
16:46:31.611 conference.c Port 4 (sip:4 at titikshu) transmitting to port 0
(HDA Intel: AD198x Analog (hw:0,0))
16:46:31.611 conference.c Port 0 (HDA Intel: AD198x Analog (hw:0,0))
transmitting to port 4 (sip:4 at titikshu)
16:46:31.611 pjsua_app.c Media for call 0 is active
16:46:31.611 pjsua_core.c TX 344 bytes Request msg ACK/cseq=26487
(tdta0x18edbaf0) to UDP 196.1.110.214:5060:
ACK sip:4 at 196.1.110.214:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.105.90:5060
;rport;branch=z9hG4bKPj097fb5ae-26f0-4cc4-9c07-1d4ec681ba7e
Max-Forwards: 70
From: sip:user1@xxxxxxxxxxx <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
To: sip:4 at titikshu;tag=1673815833
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 26487 ACK
Content-Length: 0
--end msg--
16:46:31.611 pjsua_app.c Call 0 state changed to CONFIRMED
16:46:38.531 pjsua_core.c RX 413 bytes Request msg BYE/cseq=7560
(rdata0x18e8d508) from UDP 196.1.110.214:5060:
BYE sip:user1 at 192.168.105.90:5060 SIP/2.0
Via: SIP/2.0/UDP 196.1.110.214:5060
;rport;branch=z9hG4bK6CF4B2ACDC0273EC25F0F9C7FE83DC8D
From: <sip:4@titikshu>;tag=1673815833
To: sip:user1 at example.com <sip%3Auser1 at example.com>
;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
Contact: <sip:4 at 196.1.110.214:5060>
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
CSeq: 7560 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1105d
Content-Length: 0
--end msg--
16:46:38.531 pjsua_core.c TX 328 bytes Response msg 200/BYE/cseq=7560
(tdta0x18eddab0) to UDP 196.1.110.214:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 196.1.110.214:5060
;rport=5060;received=196.1.110.214;branch=z9hG4bK6CF4B2ACDC0273EC25F0F9C7FE83DC8D
Call-ID: 70605430-8f48-4c1e-a8d5-b27331f9bb32
From: <sip:4@titikshu>;tag=1673815833
To: <sip:user1 at example.com <sip%3Auser1 at example.com>
>;tag=0b10e4a3-cc6e-452b-a427-dddf63e539e2
CSeq: 7560 BYE
Content-Length: 0
--end msg--
16:46:38.531 pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal
call clearing)]
16:46:38.531 pjsua_app.c
[DISCONNCTD] To: sip:4 at titikshu;tag=1673815833
Call time: 00h:00m:06s, 1st res in 4 ms, conn in 7035ms
SRTP status: Not active Crypto-suite: (null)
#0 iLBC @8KHz, sendrecv, peer=196.1.110.214:4000
RX pt=109, stat last update: never
total 3pkt 150B (270B +IP hdr) @avg=173bps/312bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : -0.001 0.000 0.000 0.000 0.000
TX pt=98, ptime=30ms, stat last update: 00h:00m:01.887s ago
total 1pkt 0B (40B +IP hdr) @avg 0bps/46bps
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
16:46:38.531 pjsua_media.c Media session for call 0 is destroyed
16:46:39.531 pjsua_media.c Closing sound device after idle for 1 seconds
16:46:39.531 pjsua_media.c Closing HDA Intel: AD198x Analog (hw:0,0)
sound playback device and HDA Intel: AD198x Analog (hw:0,0) sound capture
device
h
No current call
>>> q
17:02:01.787 pjsua_core.c Shutting down...
17:02:01.788 pjsua_pres.c Shutting down presence..
17:02:01.788 pjsua_media.c Shutting down media..
17:02:02.188 pa_dev.c PortAudio sound library shutting down..
17:02:03.189 pjsua_core.c Destroying...
17:02:03.189 sip_transactio Stopping transaction layer module
17:02:03.189 sip_endpoint.c Module "mod-default-handler" unregistered
17:02:03.190 sip_endpoint.c Module "mod-unsolicited-mwi" unregistered
17:02:03.190 sip_endpoint.c Module "mod-pjsua-options" unregistered
17:02:03.190 sip_endpoint.c Module "mod-pjsua-im" unregistered
17:02:03.190 sip_endpoint.c Module "mod-pjsua-pres" unregistered
17:02:03.190 sip_endpoint.c Module "mod-pjsua" unregistered
17:02:03.190 sip_endpoint.c Module "mod-stateful-util" unregistered
17:02:03.190 sip_endpoint.c Module "mod-refer" unregistered
17:02:03.190 sip_endpoint.c Module "mod-mwi" unregistered
17:02:03.190 sip_endpoint.c Module "mod-presence" unregistered
17:02:03.190 sip_endpoint.c Module "mod-evsub" unregistered
17:02:03.190 sip_endpoint.c Module "mod-invite" unregistered
17:02:03.190 sip_endpoint.c Module "mod-100rel" unregistered
17:02:03.190 sip_endpoint.c Module "mod-ua" unregistered
17:02:03.190 sip_transactio Transaction layer module destroyed
17:02:03.190 sip_endpoint.c Module "mod-tsx-layer" unregistered
17:02:03.190 sip_endpoint.c Module "mod-msg-print" unregistered
17:02:03.190 sip_endpoint.c Module "mod-pjsua-log" unregistered
17:02:03.191 tcplis:5060 SIP TCP listener destroyed
17:02:03.191 sip_endpoint.c Endpoint 0x18e31978 destroyed
17:02:03.191 sample.wav Pool is not released by application, releasing
now
17:02:03.191 pjsua_core.c PJSUA destroyed...
Kindly reply ASAP.
Thanks and Regards,
Shivani Singh Atoliya
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