URGENT - pjsip 2.1 - audio playback, audio capture do not work at all its being 4 weeks now any advise on this please?
Hello Gaurav, Varun, Dennis, Nishant
Can you please tell me why after installing pjsip 2.1 perfectly, with
libasound2 and all , i still do not have audio playback? (i checked with
speaker-test, alsa sink, src mplayer, vlc, ffmpeg my speaker and mic is
available without pjsip it works, but with pjsip i still do not hear any
single audio playback, also when i am connected i have no microphone
capture)
Please can you kindly share, i have been trying this for about now 4 weeks,
still its not working at all.
Please see the details of following steps how i installed it and how i
tested it.
Step 1: install and run
$ cd /var/tmp
$ wget http://www.pjsip.org/release/2.1/pjproject-2.1.tar.bz2
$ tar xvfj pjproject-2.1.tar.bz2
$ cd pjproject-2.1
$ ./configure
$ make dep && make && make install
Python enable (optional)
$ cd /var/tmp/pjproject-2.1.0/pjsip-apps/src/python
$ python setup.py install
$ python
Python 2.7.5+ (default, Sep 19 2013, 13:48:49)
[GCC 4.8.1] on linux2
Type "help", "copyright", "credits" or "license" for more information.
import pjsua
Step 2: Basic kick start sample to register and make call, by manually
assigning playback id and capture id , this also do not work for audio
capture and playback: https://gist.github.com/anonymous/7768285
Here you can see i used the latest release built in pjsua which also giving
no sound and no luck to capture microphone.
$ ./pjsua-x86_64-unknown-linux-gnu
13:24:33.632 os_core_unix.c !pjlib 2.1 for POSIX initialized
13:24:33.632 sip_endpoint.c .Creating endpoint instance...
13:24:33.633 pjlib .select() I/O Queue created (0x20fb8a0)
13:24:33.633 sip_endpoint.c .Module "mod-msg-print" registered
13:24:33.633 sip_transport. .Transport manager created.
13:24:33.633 pjsua_core.c .PJSUA state changed: NULL --> CREATED
13:24:33.633 sip_endpoint.c .Module "mod-pjsua-log" registered
13:24:33.633 sip_endpoint.c .Module "mod-tsx-layer" registered
13:24:33.633 sip_endpoint.c .Module "mod-stateful-util" registered
13:24:33.633 sip_endpoint.c .Module "mod-ua" registered
13:24:33.633 sip_endpoint.c .Module "mod-100rel" registered
13:24:33.633 sip_endpoint.c .Module "mod-pjsua" registered
13:24:33.633 sip_endpoint.c .Module "mod-invite" registered
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
13:24:33.702 pa_dev.c ..PortAudio sound library initialized, status=0
13:24:33.702 pa_dev.c ..PortAudio host api count=2
13:24:33.702 pa_dev.c ..Sound device count=20
13:24:33.702 pjlib ..select() I/O Queue created (0x21579f8)
13:24:33.711 sip_endpoint.c .Module "mod-evsub" registered
13:24:33.711 sip_endpoint.c .Module "mod-presence" registered
13:24:33.711 sip_endpoint.c .Module "mod-mwi" registered
13:24:33.711 sip_endpoint.c .Module "mod-refer" registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-pres" registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-im" registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-options" registered
13:24:33.711 pjsua_core.c .1 SIP worker threads created
13:24:33.711 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/
glibc-2.17 initialized
13:24:33.711 pjsua_core.c .PJSUA state changed: CREATED --> INIT
13:24:33.711 sip_endpoint.c Module "mod-default-handler" registered
13:24:33.711 pjsua_core.c bind() error: Address already in use [status=
120098]
13:24:33.711 pjsua_core.c Shutting down, flags=0...
13:24:33.711 pjsua_core.c PJSUA state changed: INIT --> CLOSING
13:24:33.721 pjsua_call.c .Hangup all calls..
13:24:33.721 pjsua_pres.c .Shutting down presence..
13:24:33.721 pjsua_media.c .Shutting down media..
13:24:33.721 pjsua_media.c ..Call 0: deinitializing media..
13:24:33.721 pjsua_media.c ..Call 1: deinitializing media..
13:24:33.721 pjsua_media.c ..Call 2: deinitializing media..
13:24:33.721 pjsua_media.c ..Call 3: deinitializing media..
13:24:34.203 pa_dev.c ..PortAudio sound library shutting down..
13:24:35.210 pjsua_core.c .Destroying...
13:24:35.210 sip_transactio .Stopping transaction layer module
13:24:35.210 sip_transactio .Stopped transaction layer module
13:24:35.210 sip_endpoint.c .Module "mod-default-handler" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-options" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-im" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-pres" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-stateful-util" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-refer" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-mwi" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-presence" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-evsub" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-invite" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-100rel" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-ua" unregistered
13:24:35.210 sip_transactio .Transaction layer module destroyed
13:24:35.210 sip_endpoint.c .Module "mod-tsx-layer" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-msg-print" unregistered
13:24:35.211 sip_endpoint.c .Module "mod-pjsua-log" unregistered
13:24:35.211 sip_endpoint.c .Endpoint 0x20f0b08 destroyed
13:24:35.211 pjsua_core.c .PJSUA state changed: CLOSING --> NULL
13:24:35.211 pjsua_core.c .PJSUA destroyed...
sun@sun-Alienware-X51:/var/tmp/pjproject-2.1.0/pjsip-apps/bin$ ./pjsua-
x86_64-unknown-linux-gnu
13:24:51.994 os_core_unix.c !pjlib 2.1 for POSIX initialized
13:24:51.995 sip_endpoint.c .Creating endpoint instance...
13:24:51.995 pjlib .select() I/O Queue created (0x9d98a0)
13:24:51.995 sip_endpoint.c .Module "mod-msg-print" registered
13:24:51.995 sip_transport. .Transport manager created.
13:24:51.995 pjsua_core.c .PJSUA state changed: NULL --> CREATED
13:24:51.995 sip_endpoint.c .Module "mod-pjsua-log" registered
13:24:51.995 sip_endpoint.c .Module "mod-tsx-layer" registered
13:24:51.995 sip_endpoint.c .Module "mod-stateful-util" registered
13:24:51.995 sip_endpoint.c .Module "mod-ua" registered
13:24:51.995 sip_endpoint.c .Module "mod-100rel" registered
13:24:51.995 sip_endpoint.c .Module "mod-pjsua" registered
13:24:51.995 sip_endpoint.c .Module "mod-invite" registered
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
13:24:52.013 pa_dev.c ..PortAudio sound library initialized, status=0
13:24:52.013 pa_dev.c ..PortAudio host api count=2
13:24:52.013 pa_dev.c ..Sound device count=20
13:24:52.013 pjlib ..select() I/O Queue created (0xa359f8)
13:24:52.016 sip_endpoint.c .Module "mod-evsub" registered
13:24:52.016 sip_endpoint.c .Module "mod-presence" registered
13:24:52.017 sip_endpoint.c .Module "mod-mwi" registered
13:24:52.017 sip_endpoint.c .Module "mod-refer" registered
13:24:52.017 sip_endpoint.c .Module "mod-pjsua-pres" registered
13:24:52.017 sip_endpoint.c .Module "mod-pjsua-im" registered
13:24:52.017 sip_endpoint.c .Module "mod-pjsua-options" registered
13:24:52.017 pjsua_core.c .1 SIP worker threads created
13:24:52.017 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/
glibc-2.17 initialized
13:24:52.017 pjsua_core.c .PJSUA state changed: CREATED --> INIT
13:24:52.017 sip_endpoint.c Module "mod-default-handler" registered
13:24:52.017 pjsua_core.c SIP UDP socket reachable at 192.168.1.19:5060
13:24:52.017 udp0xa4e6e0 SIP UDP transport started, published address is
192.168.1.19:5060
13:24:52.017 pjsua_acc.c Adding account: id=sip:192.168.1.19:5060
13:24:52.017 pjsua_acc.c .Account sip:192.168.1.19:5060 added with id
0
13:24:52.017 pjsua_acc.c Acc 0: setting online status to 1..
13:24:52.017 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.1.19:5060
13:24:52.017 pjsua_acc.c Adding account: id=<sip:192.168.1.19:5060;
transport=TCP>
13:24:52.017 pjsua_acc.c .Account sip:192.168.1.19:5060;transport=TCP
added with id 1
13:24:52.017 pjsua_acc.c Acc 1: setting online status to 1..
13:24:52.017 pjsua_core.c PJSUA state changed: INIT --> STARTING
13:24:52.017 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
13:24:52.017 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
Account list:
[ 0] sip:192.168.1.19:5060: does not register
Online status: Online
*[ 1] sip:192.168.1.19:5060;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 |
|
+-----------------------------------------------------------------------------+
| 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:9198@192.168.1.12
13:25:48.064 pjsua_call.c Making call with acc #1 to
sip:9198@192.168.1.12
13:25:48.065 pjsua_aud.c .Set sound device: capture=-1, playback=-2
13:25:48.065 pjsua_app.c ..Turning sound device ON
13:25:48.065 pjsua_aud.c ..Opening sound device PCM@16000/1/20ms
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, &hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, &hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
13:25:48.065 pjsua_app.c ..Turning sound device ON
13:25:48.065 pjsua_aud.c ..Opening sound device PCM@44100/1/20ms
13:25:48.122 ec0x9fec00 ...AEC created, clock_rate=44100,
channel=1, samples
per frame=882, tail length=200 ms, latency=100 ms
13:25:48.123 pjsua_media.c .Call 0: initializing media..
13:25:48.123 pjsua_media.c ..RTP socket reachable at 192.168.1.19:40000
13:25:48.123 pjsua_media.c ..RTCP socket reachable at 192.168.1.19:40001
13:25:48.123 pjsua_media.c ..Media index 0 selected for audio call 0
13:25:48.123 pjsua_core.c ....TX 1107 bytes Request msg INVITE/cseq=18614
(tdta0xadcbd0) to UDP 192.168.1.12:5060:
INVITE sip:9198@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
Max-Forwards: 70
From: sip:192.168.1.19;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To: sip:9198@192.168.1.12
Contact: sip:192.168.1.19:5060;ob
Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Content-Type: application/sdp
Content-Length: 475
v=0
o=- 3595062348 3595062348 IN IP4 192.168.1.19
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.1.19
b=TIAS:64000
a=rtcp:40001 IN IP4 192.168.1.19
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
--end msg--
13:25:48.123 pjsua_app.c .......Call 0 state changed to CALLING
13:25:48.124 pjsua_core.c .RX 365 bytes Response msg 100/INVITE/cseq=
18614 (rdata0xa4fd48) from UDP 192.168.1.12:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
From: sip:192.168.1.19;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To: sip:9198@192.168.1.12
Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Content-Length: 0
--end msg--
13:25:48.144 os_core_unix.c Info: possibly re-registering existing thread
13:25:48.145 pjsua_core.c .RX 882 bytes Response msg 407/INVITE/cseq=
18614 (rdata0x7f5b40002998) from UDP 192.168.1.12:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
From: sip:192.168.1.19;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To: sip:9198@192.168.1.12;tag=DZ4am8m4t08Xr
Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="192.168.1.19", nonce=
"db2a6c3c-5c29-11e3-a388-3586b66a1730", algorithm=MD5, qop="auth"
Content-Length: 0
--end msg--
13:25:48.145 pjsua_core.c ..TX 334 bytes Request msg ACK/cseq=18614 (
tdta0x7f5b400008c0) to UDP 192.168.1.12:5060:
ACK sip:9198@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
Max-Forwards: 70
From: sip:192.168.1.19;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To: sip:9198@192.168.1.12;tag=DZ4am8m4t08Xr
Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 ACK
Content-Length: 0
--end msg--
13:25:48.145 sip_auth_clien ....Unable to set auth for tdta0xadcbd0: can
not find credential for 192.168.1.19/Digest
13:25:48.145 pjsua_app.c .....Call 0 is DISCONNECTED [reason=407 (Proxy
Authentication Required)]
13:25:48.145 pjsua_app.c .....
[DISCONNCTD] To: sip:9198@192.168.1.12
Call time: 00h:00m:00s, 1st res in 23 ms, conn in 0ms
13:25:48.145 pjsua_media.c .....Call 0: deinitializing media..
13:25:49.146 pjsua_aud.c !Closing sound device after idle for 1 second(s)
13:25:49.146 pjsua_app.c .Turning sound device OFF
13:25:49.146 pjsua_aud.c .Closing HDA Intel PCH: ALC892 Analog
(hw:0,0) sound
playback device and HDA Intel PCH: ALC892 Analog (hw:0,0) sound capture
device
q
13:26:32.391 pjsua_core.c !Shutting down, flags=0...
13:26:32.391 pjsua_core.c PJSUA state changed: RUNNING --> CLOSING
13:26:32.396 pjsua_call.c .Hangup all calls..
13:26:32.396 pjsua_pres.c .Shutting down presence..
13:26:32.396 pjsua_media.c .Shutting down media..
13:26:32.396 pjsua_media.c ..Call 0: deinitializing media..
13:26:32.396 pjsua_media.c ..Call 1: deinitializing media..
13:26:32.396 pjsua_media.c ..Call 2: deinitializing media..
13:26:32.396 pjsua_media.c ..Call 3: deinitializing media..
13:26:32.524 pa_dev.c ..PortAudio sound library shutting down..
13:26:33.532 pjsua_core.c .Destroying...
13:26:33.532 sip_transactio .Stopping transaction layer module
13:26:33.532 sip_transactio .Stopped transaction layer module
13:26:33.532 sip_endpoint.c .Module "mod-default-handler" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-unsolicited-mwi" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-options" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-im" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-pres" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-stateful-util" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-refer" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-mwi" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-presence" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-evsub" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-invite" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-100rel" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-ua" unregistered
13:26:33.532 sip_transactio .Transaction layer module destroyed
13:26:33.532 sip_endpoint.c .Module "mod-tsx-layer" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-msg-print" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-log" unregistered
13:26:33.533 tcplis:5060 .SIP TCP listener destroyed
13:26:33.533 sip_endpoint.c .Endpoint 0x9ceb08 destroyed
13:26:33.533 pjsua_core.c .PJSUA state changed: CLOSING --> NULL
13:26:33.533 pjsua_core.c .PJSUA destroyed...
Thank you
Regards
Hi ,
Seems like u r nt usng any proxy server
chk
./pjsua-x86_64-unknown-linux-gnu --help is thr any option for no
registeration and still establishing call.
plz try switch
--reg-use-proxy=0 then chk whtr u r able to establish call.
Thanks,
Gaurav
On Tue, 3 Dec 2013 13:34:32 +0100, Shamun Toha Md
wrote:
Hello
Gaurav, Varun, Dennis, Nishant
Can you please tell me why after
installing pjsip 2.1 perfectly, with
libasound2 and all , i still do not
have audio playback? (i checked with
speaker-test, alsa sink, src
mplayer, vlc, ffmpeg my speaker and mic is
available without pjsip it
works, but with pjsip i still do not hear any
single audio playback, also
when i am connected i have no microphone
capture)
Please can you
kindly share, i have been trying this for about now 4 weeks,
still its
not working at all.
Please see the details of following steps how i
installed it and how i
tested it.
Step 1: install and run
$
cd /var/tmp
$ wget
$ tar xvfj
pjproject-2.1.tar.bz2
$ cd pjproject-2.1
$ ./configure
$ make dep &&
make && make install
Python enable (optional)
$ cd
/var/tmp/pjproject-2.1.0/pjsip-apps/src/python
$ python setup.py
install
$ python
Python 2.7.5+ (default, Sep 19 2013, 13:48:49)
[GCC
4.8.1] on linux2
Type "help", "copyright", "credits" or "license" for
more information.
import pjsua
Step 2: Basic kick start
sample to register and make call, by manually
assigning playback id and
capture id , this also do not work for audio
capture and playback:
Here you can see i used the
latest release built in pjsua which also giving
no sound and no luck to
capture microphone.
$ ./pjsua-x86_64-unknown-linux-gnu
13:24:33.632
os_core_unix.c !pjlib 2.1 for POSIX initialized
13:24:33.632
sip_endpoint.c .Creating endpoint instance...
13:24:33.633 pjlib
.select() I/O Queue created (0x20fb8a0)
13:24:33.633 sip_endpoint.c
.Module "mod-msg-print" registered
13:24:33.633 sip_transport. .Transport
manager created.
13:24:33.633 pjsua_core.c .PJSUA state changed: NULL -->
CREATED
13:24:33.633 sip_endpoint.c .Module "mod-pjsua-log" registered
13:24:33.633 sip_endpoint.c .Module "mod-tsx-layer" registered
13:24:33.633 sip_endpoint.c .Module "mod-stateful-util" registered
13:24:33.633 sip_endpoint.c .Module "mod-ua" registered
13:24:33.633
sip_endpoint.c .Module "mod-100rel" registered
13:24:33.633
sip_endpoint.c .Module "mod-pjsua" registered
13:24:33.633 sip_endpoint.c
.Module "mod-invite" registered
bt_audio_service_open: connect() failed:
Connection refused (111)
bt_audio_service_open: connect() failed:
Connection refused (111)
bt_audio_service_open: connect() failed:
Connection refused (111)
bt_audio_service_open: connect() failed:
Connection refused (111)
13:24:33.702 pa_dev.c ..PortAudio sound library
initialized,
status=0
13:24:33.702 pa_dev.c ..PortAudio host api
count=2
13:24:33.702 pa_dev.c ..Sound device count=20
13:24:33.702
pjlib ..select() I/O Queue created (0x21579f8)
13:24:33.711
sip_endpoint.c .Module "mod-evsub" registered
13:24:33.711 sip_endpoint.c
.Module "mod-presence" registered
13:24:33.711 sip_endpoint.c .Module
"mod-mwi" registered
13:24:33.711 sip_endpoint.c .Module "mod-refer"
registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-pres"
registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-im"
registered
13:24:33.711 sip_endpoint.c .Module "mod-pjsua-options"
registered
13:24:33.711 pjsua_core.c .1 SIP worker threads created
13:24:33.711 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/
glibc-2.17 initialized
13:24:33.711 pjsua_core.c .PJSUA state changed:
CREATED --> INIT
13:24:33.711 sip_endpoint.c Module "mod-default-handler"
registered
13:24:33.711 pjsua_core.c bind() error: Address already in use
[status=
120098]
13:24:33.711 pjsua_core.c Shutting down, flags=0...
13:24:33.711 pjsua_core.c PJSUA state changed: INIT --> CLOSING
13:24:33.721 pjsua_call.c .Hangup all calls..
13:24:33.721 pjsua_pres.c
.Shutting down presence..
13:24:33.721 pjsua_media.c .Shutting down
media..
13:24:33.721 pjsua_media.c ..Call 0: deinitializing media..
13:24:33.721 pjsua_media.c ..Call 1: deinitializing media..
13:24:33.721
pjsua_media.c ..Call 2: deinitializing media..
13:24:33.721 pjsua_media.c
..Call 3: deinitializing media..
13:24:34.203 pa_dev.c ..PortAudio sound
library shutting down..
13:24:35.210 pjsua_core.c .Destroying...
13:24:35.210 sip_transactio .Stopping transaction layer module
13:24:35.210 sip_transactio .Stopped transaction layer module
13:24:35.210 sip_endpoint.c .Module "mod-default-handler" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-options" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-im" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-pres" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-pjsua" unregistered
13:24:35.210
sip_endpoint.c .Module "mod-stateful-util" unregistered
13:24:35.210
sip_endpoint.c .Module "mod-refer" unregistered
13:24:35.210
sip_endpoint.c .Module "mod-mwi" unregistered
13:24:35.210 sip_endpoint.c
.Module "mod-presence" unregistered
13:24:35.210 sip_endpoint.c .Module
"mod-evsub" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-invite"
unregistered
13:24:35.210 sip_endpoint.c .Module "mod-100rel"
unregistered
13:24:35.210 sip_endpoint.c .Module "mod-ua" unregistered
13:24:35.210 sip_transactio .Transaction layer module destroyed
13:24:35.210 sip_endpoint.c .Module "mod-tsx-layer" unregistered
13:24:35.210 sip_endpoint.c .Module "mod-msg-print" unregistered
13:24:35.211 sip_endpoint.c .Module "mod-pjsua-log" unregistered
13:24:35.211 sip_endpoint.c .Endpoint 0x20f0b08 destroyed
13:24:35.211
pjsua_core.c .PJSUA state changed: CLOSING --> NULL
13:24:35.211
pjsua_core.c .PJSUA destroyed...
sun@sun-Alienware-X51:/var/tmp/pjproject-2.1.0/pjsip-apps/bin$ ./pjsua-
x86_64-unknown-linux-gnu
13:24:51.994 os_core_unix.c !pjlib 2.1 for POSIX
initialized
13:24:51.995 sip_endpoint.c .Creating endpoint instance...
13:24:51.995 pjlib .select() I/O Queue created (0x9d98a0)
13:24:51.995
sip_endpoint.c .Module "mod-msg-print" registered
13:24:51.995
sip_transport. .Transport manager created.
13:24:51.995 pjsua_core.c
.PJSUA state changed: NULL --> CREATED
13:24:51.995 sip_endpoint.c
.Module "mod-pjsua-log" registered
13:24:51.995 sip_endpoint.c .Module
"mod-tsx-layer" registered
13:24:51.995 sip_endpoint.c .Module
"mod-stateful-util" registered
13:24:51.995 sip_endpoint.c .Module
"mod-ua" registered
13:24:51.995 sip_endpoint.c .Module "mod-100rel"
registered
13:24:51.995 sip_endpoint.c .Module "mod-pjsua" registered
13:24:51.995 sip_endpoint.c .Module "mod-invite" registered
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
13:24:52.013 pa_dev.c ..PortAudio sound library initialized,
status=0
13:24:52.013 pa_dev.c ..PortAudio host api count=2
13:24:52.013 pa_dev.c
..Sound device count=20
13:24:52.013 pjlib ..select() I/O Queue created
(0xa359f8)
13:24:52.016 sip_endpoint.c .Module "mod-evsub" registered
13:24:52.016 sip_endpoint.c .Module "mod-presence" registered
13:24:52.017 sip_endpoint.c .Module "mod-mwi" registered
13:24:52.017
sip_endpoint.c .Module "mod-refer" registered
13:24:52.017 sip_endpoint.c
.Module "mod-pjsua-pres" registered
13:24:52.017 sip_endpoint.c .Module
"mod-pjsua-im" registered
13:24:52.017 sip_endpoint.c .Module
"mod-pjsua-options" registered
13:24:52.017 pjsua_core.c .1 SIP worker
threads created
13:24:52.017 pjsua_core.c .pjsua version 2.1 for
Linux-3.11.0.12/x86_64/
glibc-2.17 initialized
13:24:52.017
pjsua_core.c .PJSUA state changed: CREATED --> INIT
13:24:52.017
sip_endpoint.c Module "mod-default-handler" registered
13:24:52.017
pjsua_core.c SIP UDP socket reachable at 192.168.1.19:5060
13:24:52.017
udp0xa4e6e0 SIP UDP transport started, published address
is
192.168.1.19:5060
13:24:52.017 pjsua_acc.c Adding account: id=
13:24:52.017 pjsua_acc.c .Account added with id
0
13:24:52.017
pjsua_acc.c Acc 0: setting online status to 1..
13:24:52.017 tcplis:5060
SIP TCP listener ready for incoming
connections at 192.168.1.19:5060
13:24:52.017 pjsua_acc.c Adding account: id= transport=TCP>
13:24:52.017
pjsua_acc.c .Account
added with id 1
13:24:52.017 pjsua_acc.c Acc 1:
setting online status to 1..
13:24:52.017 pjsua_core.c PJSUA state
changed: INIT --> STARTING
13:24:52.017 sip_endpoint.c .Module
"mod-unsolicited-mwi" registered
13:24:52.017 pjsua_core.c .PJSUA state
changed: STARTING --> RUNNING
Account list:
[ 0] : does not
register
Online status: Online
*[ 1] : 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 |
|
+-----------------------------------------------------------------------------+
| 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
Empty
input (or 'q') to cancel
Make call: sip:9198@192.168.1.12
13:25:48.064
pjsua_call.c Making call with acc #1 to
13:25:48.065 pjsua_aud.c .Set sound device: capture=-1, playback=-2
13:25:48.065 pjsua_app.c ..Turning sound device ON
13:25:48.065
pjsua_aud.c ..Opening sound device PCM@16000/1/20ms
Expression
'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, &hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, rport;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
Max-Forwards: 70
From: ;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To:
Contact:
Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
Allow: PRACK,
INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.1
Linux-3.11.0.12/x86_64/glibc-2.17
Content-Type: application/sdp
Content-Length: 475
v=0
o=- 3595062348 3595062348 IN IP4
192.168.1.19
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 40000
RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.1.19
b=TIAS:64000
a=rtcp:40001 IN IP4 192.168.1.19
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96
telephone-event/8000
a=fmtp:96 0-15
--end msg--
13:25:48.123
pjsua_app.c .......Call 0 state changed to CALLING
13:25:48.124
pjsua_core.c .RX 365 bytes Response msg
100/INVITE/cseq=
18614
(rdata0xa4fd48) from UDP 192.168.1.12:5060:
SIP/2.0 100 Trying
Via:
SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To:
Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
User-Agent:
FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Content-Length: 0
--end msg--
13:25:48.144 os_core_unix.c
Info: possibly re-registering existing thread
13:25:48.145 pjsua_core.c
.RX 882 bytes Response msg 407/INVITE/cseq=
18614 (rdata0x7f5b40002998)
from UDP 192.168.1.12:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To: ;tag=DZ4am8m4t08Xr
Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 INVITE
User-Agent:
FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Accept:
application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer,
precondition, path, replaces
Allow-Events: talk, hold, conference,
presence, as-feature-event, dialog,
line-seize, call-info, sla,
include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="192.168.1.19", nonce=
"db2a6c3c-5c29-11e3-a388-3586b66a1730", algorithm=MD5, qop="auth"
Content-Length: 0
--end msg--
13:25:48.145 pjsua_core.c
..TX 334 bytes Request msg ACK/cseq=18614 (
tdta0x7f5b400008c0) to UDP
192.168.1.12:5060:
ACK sip:9198@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.19:5060;rport;branch=
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
Max-Forwards: 70
From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
To:
sip:9198@192.168.1.12;tag=DZ4am8m4t08Xr
Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
CSeq: 18614 ACK
Content-Length: 0
--end msg--
13:25:48.145 sip_auth_clien ....Unable to set auth
for tdta0xadcbd0: can
not find credential for 192.168.1.19/Digest
13:25:48.145 pjsua_app.c .....Call 0 is DISCONNECTED [reason=407 (Proxy
Authentication Required)]
13:25:48.145 pjsua_app.c .....
[DISCONNCTD]
To: sip:9198@192.168.1.12
Call time: 00h:00m:00s, 1st res in 23 ms, conn
in 0ms
13:25:48.145 pjsua_media.c .....Call 0: deinitializing media..
13:25:49.146 pjsua_aud.c !Closing sound device after idle for 1
second(s)
13:25:49.146 pjsua_app.c .Turning sound device OFF
13:25:49.146 pjsua_aud.c .Closing HDA Intel PCH: ALC892 Analog
(hw:0,0)
sound
playback device and HDA Intel PCH: ALC892 Analog (hw:0,0) sound
capture
device
q
13:26:32.391 pjsua_core.c !Shutting
down, flags=0...
13:26:32.391 pjsua_core.c PJSUA state changed: RUNNING
--> CLOSING
13:26:32.396 pjsua_call.c .Hangup all calls..
13:26:32.396
pjsua_pres.c .Shutting down presence..
13:26:32.396 pjsua_media.c
.Shutting down media..
13:26:32.396 pjsua_media.c ..Call 0:
deinitializing media..
13:26:32.396 pjsua_media.c ..Call 1:
deinitializing media..
13:26:32.396 pjsua_media.c ..Call 2:
deinitializing media..
13:26:32.396 pjsua_media.c ..Call 3:
deinitializing media..
13:26:32.524 pa_dev.c ..PortAudio sound library
shutting down..
13:26:33.532 pjsua_core.c .Destroying...
13:26:33.532
sip_transactio .Stopping transaction layer module
13:26:33.532
sip_transactio .Stopped transaction layer module
13:26:33.532
sip_endpoint.c .Module "mod-default-handler" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-unsolicited-mwi" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-pjsua-options" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-pjsua-im" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-pjsua-pres" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-pjsua" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-stateful-util" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-refer" unregistered
13:26:33.532
sip_endpoint.c .Module "mod-mwi" unregistered
13:26:33.532 sip_endpoint.c
.Module "mod-presence" unregistered
13:26:33.532 sip_endpoint.c .Module
"mod-evsub" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-invite"
unregistered
13:26:33.532 sip_endpoint.c .Module "mod-100rel"
unregistered
13:26:33.532 sip_endpoint.c .Module "mod-ua" unregistered
13:26:33.532 sip_transactio .Transaction layer module destroyed
13:26:33.532 sip_endpoint.c .Module "mod-tsx-layer" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-msg-print" unregistered
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-log" unregistered
13:26:33.533 tcplis:5060 .SIP TCP listener destroyed
13:26:33.533
sip_endpoint.c .Endpoint 0x9ceb08 destroyed
13:26:33.533 pjsua_core.c
.PJSUA state changed: CLOSING --> NULL
13:26:33.533 pjsua_core.c .PJSUA
destroyed...
Thank you
Regards
Hello Gaurav,
YES - basically when you call SIP server without being registered you can
still hear IVR messages like invalid or something else. I do not hear any
sound.
i also tried to register and still same no sound.
You can register and then call also:
$ pjsua-x86_64-unknown-linux-gnu
--id=sip:1@192.168.1.12--registrar=sip:192.168.1.12 --realm="*"
--username=1 --password=admin2013
You can also tell which capture device and which playback device to use:
$ pjsua-x86_64-unknown-linux-gnu
--id=sip:1@192.168.1.12--registrar=sip:192.168.1.12 --realm="*"
--username=1 --password=admin2013
--capture-dev=5 --playback-dev=0
But what-ever is given i tried and i do not have audio output and audio
capture working.
Why nobody knows anything about this issue???? i have tried version 0.9
till 2.1 all have same issue with me.
Any advise fix for this plz.
Thank you
Reg
Hello *Andreas, *Gaurav, Varun, Dennis, Nishant, All,
What is the update for this issue please? No audio for playback and no
audio source captured.
Thank you
Regards