How to play a "splash-tone" file as soon as posssible in the call sequence
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard drive to the Conference Bridge port should be transmitted to the VoIP phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better respond. I can see you're answering the call and connecting the player, so those parts look correct, but it's hard to know if you are missing something important in initialization. Do you have a logfile for the auto-answer call? It should show whether the call is properly connected and may have clues as to why the player isn't playing. For example, there may be a problem with audio device that is preventing media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for how to cause a "PleaseLeaveYourMessage auto-play to happen, a few things:
1: my app (running on ARM7) receives my test call from a company digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp from the auto-play example, here's what I think should work but doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS) SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) != PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(), &mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage, PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS) SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL, &on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg, NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) != PJ_SUCCESS) SleepForever("Can't create account (you are a no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't start pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you don't want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex < CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo, CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but I can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins (hint/beg of benny: please number the tests and print out the number so I can say "test 4373 failed; it said "foobar" when it should have said "mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769 (tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM 129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset by peer)
20:02:58.932 State changed from Completed to Terminated, event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to conference when call answers. It's all part of the --auto-play option in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me correct code?
Given a call_id, how do I tell which conference port to connect to, in order to send a wav file to the caller? So far I've managed to connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call sequence wherein I'm able to tell which conference port to connect my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and therefore it is the code that is broken, not its location) lets look at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Creating a player and connecting to the conference should work, perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash tone" or "please leave your message after the tone" file from the hard drive to the caller?
I create a player, and I connect it to the conference, but clearly not the right way or right time.
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The audio device, while in this case is not used to playback or record,
is used to clock media flow. You could alternatively set up pjsip to use
the null-audio device which is just a timer. In your log I see the audio
device is initialized OK:
15:45:40.528 pjsua_aud.c .......Opening sound device PCM@16000/1/20ms
15:45:40.537 pa_dev.c ........Opened device imx6q-qmx6-sgtl5000:
(hw:0,0)(ALSA)/imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA) for recording and
playback, sample rate=16000, ch=1, bits=16, 320 samples per frame, input
latency=100 ms, output latency=140 ms
And after call connects the player is started:
15:45:40.549 pa_dev.c !Recorder thread started
15:45:40.554 os_core_unix.c Info: possibly re-registering existing thread
15:45:40.554 pa_dev.c !Player thread started
15:45:40.560 strm0x7570fc8c Start talksprut..
If memory serves, "Start talksprut..." indicates the voice detector has
detected voice onset, so this indicates signal is flowing. It all looks
good.
Next step: capture wireshark trace and see if media is sent on RTP?
Bill
On 10/31/2014 9:01 AM, Brunner, Brian T. wrote:
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard
drive to the Conference Bridge port should be transmitted to the VoIP
phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better respond.
I can see you're answering the call and connecting the player, so
those parts look correct, but it's hard to know if you are missing
something important in initialization. Do you have a logfile for the
auto-answer call? It should show whether the call is properly
connected and may have clues as to why the player isn't playing. For
example, there may be a problem with audio device that is preventing
media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for
how to cause a "PleaseLeaveYourMessage auto-play to happen, a few things:
1: my app (running on ARM7) receives my test call from a company
digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp
from the auto-play example, here's what I think should work but
doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't
create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS)
SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) !=
PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(),
&mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register
Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage,
PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS)
SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL,
&on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof
cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg,
NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad
config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) !=
PJ_SUCCESS) SleepForever("Can't create account (you are a
no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't start
pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you don't
want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex <
CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo,
CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial
tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but I
can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an
error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins (hint/beg
of benny: please number the tests and print out the number so I can
say "test 4373 failed; it said "foobar" when it should have said
"mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769
(tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM
129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg
200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset by
peer)
20:02:58.932 State changed from Completed to Terminated,
event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769
(tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to
conference when call answers. It's all part of the --auto-play option
in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me
correct code?
Given a call_id, how do I tell which conference port to connect to,
in order to send a wav file to the caller? So far I've managed to
connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call
sequence wherein I'm able to tell which conference port to connect
my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and
therefore it is the code that is broken, not its location) lets look
at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Creating a player and connecting to the conference should work,
perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash tone"
or "please leave your message after the tone" file from the hard
drive to the caller?
I create a player, and I connect it to the conference, but clearly
not the right way or right time.
Confidentiality Requirement: This communication, including any
attachment(s), may contain confidential information and is for the
sole use of the intended recipient(s). If you are not the intended
recipient, you are hereby notified that you have received this
communication in error and any unauthorized review, use,
disclosure, dissemination, distribution or copying of it or its
contents is strictly prohibited. If you have received this
communication in error, please notify the sender immediately by
telephone or e-mail and destroy all copies of this communication
and any attachments.
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog:http://blog.pjsip.org
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<quote>Next step: capture wireshark trace and see if media is sent on RTP?</quote>
Good call: According to WireShark, RTP is between my target (.229) and the ip-pbx box (.141, which is my gateway to the caller).
The first several RTP packets go both ways, then it's from ip-pbx only.
No. Time Source Destination Protocol Length Info
579 28.662739000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20333, Time=160, Mark
Frame 579: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
580 28.673127000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20334, Time=320
Frame 580: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
581 28.683403000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20335, Time=480
Frame 581: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
582 28.693534000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20336, Time=640
Frame 582: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
583 28.703727000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20337, Time=800
Frame 583: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
584 28.704927000 ip-pbx.bigman.com 192.168.1.229 SIP 610 Request: ACK sip:173@192.168.1.229:5060 |
Frame 584: 610 bytes on wire (4880 bits), 610 bytes captured (4880 bits) on interface 0
Ethernet II, Src: Dell_1e:d1:21 (00:24:e8:1e:d1:21), Dst: Congatec_0f:f5:6b (00:13:95:0f:f5:6b)
Internet Protocol Version 4, Src: ip-pbx.bigman.com (192.168.1.141), Dst: 192.168.1.229 (192.168.1.229)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
No. Time Source Destination Protocol Length Info
588 28.718298000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20338, Time=960
Frame 588: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
591 28.731045000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20339, Time=1120
Frame 591: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
592 28.752129000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20340, Time=1280
Frame 592: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
597 28.766290000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20341, Time=1440
Frame 597: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
598 28.771482000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20342, Time=1600
Frame 598: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
599 28.776671000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20343, Time=1760
Frame 599: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
600 28.781771000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20344, Time=1920
Frame 600: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
613 30.407229000 192.168.1.229 ip-pbx.bigman.com UDP 60 Source port: sip Destination port: sip
Frame 613: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Data (2 bytes)
NOTE: after 1.74 seconds of bi-directional RTP, (from 28.66 seconds to 30.40 seconds) RTP from .229 to ip-pbx ceases, and all further RTP is only from ip-pbx to .229.
RTP messages come from ip-pbx every 20ms except for a 190ms silence at the end of the 229->141 RTP.
There is one RTCP message from .229 at 30.77
The whole conversation ends 7 seconds after it starts when .229 sends BYE as it is supposed to.
NOTE: The "PleaseLeaveYourMessage.wav" file is 14MB, lasting 80 seconds. I don't hear any of it.
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.com]
Sent: Friday, October 31, 2014 10:10 AM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
The audio device, while in this case is not used to playback or record, is used to clock media flow. You could alternatively set up pjsip to use the null-audio device which is just a timer. In your log I see the audio device is initialized OK:
15:45:40.528 pjsua_aud.c .......Opening sound device PCM@16000/1/20ms
15:45:40.537 pa_dev.c ........Opened device imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA)/imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA) for recording and playback, sample rate=16000, ch=1, bits=16, 320 samples per frame, input latency=100 ms, output latency=140 ms
And after call connects the player is started:
15:45:40.549 pa_dev.c !Recorder thread started
15:45:40.554 os_core_unix.c Info: possibly re-registering existing thread
15:45:40.554 pa_dev.c !Player thread started
15:45:40.560 strm0x7570fc8c Start talksprut..
If memory serves, "Start talksprut..." indicates the voice detector has detected voice onset, so this indicates signal is flowing. It all looks good.
Next step: capture wireshark trace and see if media is sent on RTP?
Bill
On 10/31/2014 9:01 AM, Brunner, Brian T. wrote:
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard drive to the Conference Bridge port should be transmitted to the VoIP phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better respond. I can see you're answering the call and connecting the player, so those parts look correct, but it's hard to know if you are missing something important in initialization. Do you have a logfile for the auto-answer call? It should show whether the call is properly connected and may have clues as to why the player isn't playing. For example, there may be a problem with audio device that is preventing media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for how to cause a "PleaseLeaveYourMessage auto-play to happen, a few things:
1: my app (running on ARM7) receives my test call from a company digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp from the auto-play example, here's what I think should work but doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS) SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) != PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(), &mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage, PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS) SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL, &on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg, NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) != PJ_SUCCESS) SleepForever("Can't create account (you are a no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't start pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you don't want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex < CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo, CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but I can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins (hint/beg of benny: please number the tests and print out the number so I can say "test 4373 failed; it said "foobar" when it should have said "mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769 (tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM 129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset by peer)
20:02:58.932 State changed from Completed to Terminated, event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to conference when call answers. It's all part of the --auto-play option in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me correct code?
Given a call_id, how do I tell which conference port to connect to, in order to send a wav file to the caller? So far I've managed to connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call sequence wherein I'm able to tell which conference port to connect my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and therefore it is the code that is broken, not its location) lets look at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Creating a player and connecting to the conference should work, perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash tone" or "please leave your message after the tone" file from the hard drive to the caller?
I create a player, and I connect it to the conference, but clearly not the right way or right time.
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Some ideas. Using wireshark you can dump the RTP contents of that stream
from .141 and convert it to raw PCMU (using some rtp utils from
columbia, can't remember name) and then import raw pcmu into a sound
editor to play. There may be an easier way too but that's how I've done
it. That could be revealing.
The 7 sec duration is due to a call timer expiring. When call ends the
media stats dump is very useful to see but it isn't in your log, maybe
you need to increase log level.
Your WAV is 48k stereo, I assume 16-bits. Should be fine as pjsip
converts automatically. You might want to try converting it to 16k mono
16-bit just to be sure the WAV format isn't the issue.
That's all I've got.
Bill
On 10/31/2014 3:58 PM, Brunner, Brian T. wrote:
<quote>Next step: capture wireshark trace and see if media is sent on
RTP?</quote>
Good call: According to WireShark, RTP is between my target (.229) and
the ip-pbx box (.141, which is my gateway to the caller).
The first several RTP packets go both ways, then it's from ip-pbx only.
No. Time Source Destination Protocol Length Info
579 28.662739000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20333, Time=160, Mark
Frame 579: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
580 28.673127000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20334, Time=320
Frame 580: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
581 28.683403000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20335, Time=480
Frame 581: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
582 28.693534000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20336, Time=640
Frame 582: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
583 28.703727000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20337, Time=800
Frame 583: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
584 28.704927000 ip-pbx.bigman.com 192.168.1.229
SIP 610 Request: ACK sip:173@192.168.1.229:5060 |
Frame 584: 610 bytes on wire (4880 bits), 610 bytes captured (4880
bits) on interface 0
Ethernet II, Src: Dell_1e:d1:21 (00:24:e8:1e:d1:21), Dst:
Congatec_0f:f5:6b (00:13:95:0f:f5:6b)
Internet Protocol Version 4, Src: ip-pbx.bigman.com (192.168.1.141),
Dst: 192.168.1.229 (192.168.1.229)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
No. Time Source Destination Protocol Length Info
588 28.718298000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20338, Time=960
Frame 588: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
591 28.731045000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20339, Time=1120
Frame 591: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
592 28.752129000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20340, Time=1280
Frame 592: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
597 28.766290000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20341, Time=1440
Frame 597: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
598 28.771482000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20342, Time=1600
Frame 598: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
599 28.776671000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20343, Time=1760
Frame 599: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
600 28.781771000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20344, Time=1920
Frame 600: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
613 30.407229000 192.168.1.229 ip-pbx.bigman.com UDP
60 Source port: sip Destination port: sip
Frame 613: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Data (2 bytes)
NOTE: after 1.74 seconds of bi-directional RTP, (from 28.66 seconds to
30.40 seconds) RTP from .229 to ip-pbx ceases, and all further RTP is
only from ip-pbx to .229.
RTP messages come from ip-pbx every 20ms except for a 190ms silence at
the end of the 229->141 RTP.
There is one RTCP message from .229 at 30.77
The whole conversation ends 7 seconds after it starts when .229 sends
BYE as it is supposed to.
NOTE: The "PleaseLeaveYourMessage.wav" file is 14MB, lasting 80
seconds. I don't hear any of it.
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 31, 2014 10:10 AM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
The audio device, while in this case is not used to playback or
record, is used to clock media flow. You could alternatively set up
pjsip to use the null-audio device which is just a timer. In your log
I see the audio device is initialized OK:
15:45:40.528 pjsua_aud.c .......Opening sound device PCM@16000/1/20ms
15:45:40.537 pa_dev.c ........Opened device
imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA)/imx6q-qmx6-sgtl5000:
(hw:0,0)(ALSA) for recording and playback, sample rate=16000, ch=1,
bits=16, 320 samples per frame, input latency=100 ms, output
latency=140 ms
And after call connects the player is started:
15:45:40.549 pa_dev.c !Recorder thread started
15:45:40.554 os_core_unix.c Info: possibly re-registering existing thread
15:45:40.554 pa_dev.c !Player thread started
15:45:40.560 strm0x7570fc8c Start talksprut..
If memory serves, "Start talksprut..." indicates the voice detector
has detected voice onset, so this indicates signal is flowing. It all
looks good.
Next step: capture wireshark trace and see if media is sent on RTP?
Bill
On 10/31/2014 9:01 AM, Brunner, Brian T. wrote:
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard
drive to the Conference Bridge port should be transmitted to the VoIP
phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better respond.
I can see you're answering the call and connecting the player, so
those parts look correct, but it's hard to know if you are missing
something important in initialization. Do you have a logfile for the
auto-answer call? It should show whether the call is properly
connected and may have clues as to why the player isn't playing. For
example, there may be a problem with audio device that is preventing
media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for
how to cause a "PleaseLeaveYourMessage auto-play to happen, a few
things:
1: my app (running on ARM7) receives my test call from a company
digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp
from the auto-play example, here's what I think should work but
doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't
create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS)
SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) !=
PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(),
&mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register
Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage,
PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS)
SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL,
&on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof
cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg,
NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad
config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) !=
PJ_SUCCESS) SleepForever("Can't create account (you are a
no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't
start pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you
don't want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex <
CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo,
CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial
tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but I
can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an
error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins
(hint/beg of benny: please number the tests and print out the number
so I can say "test 4373 failed; it said "foobar" when it should have
said "mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769
(tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM
129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg
200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset
by peer)
20:02:58.932 State changed from Completed to Terminated,
event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769
(tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to
conference when call answers. It's all part of the --auto-play
option in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me
correct code?
Given a call_id, how do I tell which conference port to connect to,
in order to send a wav file to the caller? So far I've managed to
connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call
sequence wherein I'm able to tell which conference port to connect
my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and
therefore it is the code that is broken, not its location) lets
look at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Creating a player and connecting to the conference should work,
perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash tone"
or "please leave your message after the tone" file from the hard
drive to the caller?
I create a player, and I connect it to the conference, but clearly
not the right way or right time.
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Is there any clear documentation describing which conference bridge slot does what?
Is there any debug setting by which I can tell whether the SplashTone file is being read and routed to the callers' conference bridge slot?
From my reading the documentation, "somebody" must be putting frames on the file side, and getting frames on the callers' side, in order for the audio data to get to the phone.
I don't have a "somebody()" routine, is pjsua/pjmedia (supposedly) doing this?
There are only a few rtp packets originating from my side.
Media Stats:
[DISCONNCTD] To: "aa2" sip:5@192.168.1.254;tag=be69e5f44p
Call time: 00h:00m:06s, 1st res in 25 ms, conn in 130ms
#0 audio PCMU @8kHz, sendrecv, peer=192.168.1.141:11834
SRTP status: Not active Crypto-suite:
RX pt=0, last update:00h:00m:04.558s ago
total 237pkt 37.9KB (47.4KB +IP hdr) @avg=45.0Kbps/56.3Kbps
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 : 0.000 0.366 3.375 0.125 0.635
TX pt=0, ptime=20, last update:never
total 84pkt 13.4KB (16.8KB +IP hdr) @avg=15.9Kbps/19.9Kbps
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
and
strm0x73d0fc8c .......JB summary:
size=50/eff=50 prefetch=0 level=0
delay (min/max/avg/dev)=0/0/0/0 ms
burst (min/max/avg/dev)=0/0/0/0 frames
lost=0 discard=424 empty=84
I'm guessing this means I never sent a single buffer of audio to the caller.
I'll try the audio editor method to determine whether the caller sent voice data.
No data was written to the .wav file
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.com]
Sent: Friday, October 31, 2014 5:02 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Some ideas. Using wireshark you can dump the RTP contents of that stream from .141 and convert it to raw PCMU (using some rtp utils from columbia, can't remember name) and then import raw pcmu into a sound editor to play. There may be an easier way too but that's how I've done it. That could be revealing.
The 7 sec duration is due to a call timer expiring. When call ends the media stats dump is very useful to see but it isn't in your log, maybe you need to increase log level.
Your WAV is 48k stereo, I assume 16-bits. Should be fine as pjsip converts automatically. You might want to try converting it to 16k mono 16-bit just to be sure the WAV format isn't the issue.
That's all I've got.
Bill
On 10/31/2014 3:58 PM, Brunner, Brian T. wrote:
<quote>Next step: capture wireshark trace and see if media is sent on RTP?</quote>
Good call: According to WireShark, RTP is between my target (.229) and the ip-pbx box (.141, which is my gateway to the caller).
The first several RTP packets go both ways, then it's from ip-pbx only.
No. Time Source Destination Protocol Length Info
579 28.662739000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20333, Time=160, Mark
Frame 579: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
580 28.673127000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20334, Time=320
Frame 580: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
581 28.683403000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20335, Time=480
Frame 581: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
582 28.693534000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20336, Time=640
Frame 582: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
583 28.703727000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20337, Time=800
Frame 583: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
584 28.704927000 ip-pbx.bigman.com 192.168.1.229 SIP 610 Request: ACK sip:173@192.168.1.229:5060mailto:sip:173@192.168.1.229:5060 |
Frame 584: 610 bytes on wire (4880 bits), 610 bytes captured (4880 bits) on interface 0
Ethernet II, Src: Dell_1e:d1:21 (00:24:e8:1e:d1:21), Dst: Congatec_0f:f5:6b (00:13:95:0f:f5:6b)
Internet Protocol Version 4, Src: ip-pbx.bigman.com (192.168.1.141), Dst: 192.168.1.229 (192.168.1.229)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
No. Time Source Destination Protocol Length Info
588 28.718298000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20338, Time=960
Frame 588: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
591 28.731045000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20339, Time=1120
Frame 591: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
592 28.752129000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20340, Time=1280
Frame 592: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
597 28.766290000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20341, Time=1440
Frame 597: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
598 28.771482000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20342, Time=1600
Frame 598: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
599 28.776671000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20343, Time=1760
Frame 599: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
600 28.781771000 192.168.1.229 ip-pbx.bigman.com RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20344, Time=1920
Frame 600: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432 (11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
613 30.407229000 192.168.1.229 ip-pbx.bigman.com UDP 60 Source port: sip Destination port: sip
Frame 613: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst: Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst: ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Data (2 bytes)
NOTE: after 1.74 seconds of bi-directional RTP, (from 28.66 seconds to 30.40 seconds) RTP from .229 to ip-pbx ceases, and all further RTP is only from ip-pbx to .229.
RTP messages come from ip-pbx every 20ms except for a 190ms silence at the end of the 229->141 RTP.
There is one RTCP message from .229 at 30.77
The whole conversation ends 7 seconds after it starts when .229 sends BYE as it is supposed to.
NOTE: The "PleaseLeaveYourMessage.wav" file is 14MB, lasting 80 seconds. I don't hear any of it.
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Friday, October 31, 2014 10:10 AM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
The audio device, while in this case is not used to playback or record, is used to clock media flow. You could alternatively set up pjsip to use the null-audio device which is just a timer. In your log I see the audio device is initialized OK:
15:45:40.528 pjsua_aud.c .......Opening sound device PCM@16000/1/20ms
15:45:40.537 pa_dev.c ........Opened device imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA)/imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA) for recording and playback, sample rate=16000, ch=1, bits=16, 320 samples per frame, input latency=100 ms, output latency=140 ms
And after call connects the player is started:
15:45:40.549 pa_dev.c !Recorder thread started
15:45:40.554 os_core_unix.c Info: possibly re-registering existing thread
15:45:40.554 pa_dev.c !Player thread started
15:45:40.560 strm0x7570fc8c Start talksprut..
If memory serves, "Start talksprut..." indicates the voice detector has detected voice onset, so this indicates signal is flowing. It all looks good.
Next step: capture wireshark trace and see if media is sent on RTP?
Bill
On 10/31/2014 9:01 AM, Brunner, Brian T. wrote:
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard drive to the Conference Bridge port should be transmitted to the VoIP phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better respond. I can see you're answering the call and connecting the player, so those parts look correct, but it's hard to know if you are missing something important in initialization. Do you have a logfile for the auto-answer call? It should show whether the call is properly connected and may have clues as to why the player isn't playing. For example, there may be a problem with audio device that is preventing media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for how to cause a "PleaseLeaveYourMessage auto-play to happen, a few things:
1: my app (running on ARM7) receives my test call from a company digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp from the auto-play example, here's what I think should work but doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS) SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) != PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(), &mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage, PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS) SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL, &on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg, NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) != PJ_SUCCESS) SleepForever("Can't create account (you are a no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't start pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you don't want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex < CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo, CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but I can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins (hint/beg of benny: please number the tests and print out the number so I can say "test 4373 failed; it said "foobar" when it should have said "mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769 (tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769 (tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM 129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset by peer)
20:02:58.932 State changed from Completed to Terminated, event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769 (tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to conference when call answers. It's all part of the --auto-play option in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me correct code?
Given a call_id, how do I tell which conference port to connect to, in order to send a wav file to the caller? So far I've managed to connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call sequence wherein I'm able to tell which conference port to connect my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and therefore it is the code that is broken, not its location) lets look at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.orgmailto:pjsip-bounces@lists.pjsip.org] on behalf of Bill Gardner [billg@wavearts.commailto:billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as posssible in the call sequence
Creating a player and connecting to the conference should work, perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash tone" or "please leave your message after the tone" file from the hard drive to the caller?
I create a player, and I connect it to the conference, but clearly not the right way or right time.
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I'd recommend just looking at the conference.c code. The ports are
general purpose, however port 0 is always the master sound interface or
the null sound device which provides clocking to the whole mechanism.
Everything runs off of the put_frame call when port 0 need a frame to
output. The code iterates through all the ports and reads the input
samples for each port, then runs through the ports producing mixed
output samples depending on the connections, and transmits the mixed
results to the ports, then returns the mixed results to port 0. It's
something like that, going from memory.
In your case 84 packets are sent which is less than your WAV file
contains, but definitely better than 0. The fact it sent some packets
means the call is connected to an audio producing port, could be WAV
player or sound device. How to debug? Wireshark as I mentioned earlier
would let you know if those packets contain snippets of audio from your
WAV file. Or run pjsua without call but just connect port 0 to port 0
and verify that audio loops from microphone to speaker. Or connect WAV
to port 0 and see if it plays to speaker. Or add PJ_LOG to WAV player to
log individual writes. Lots of options.
Bill
On 11/6/2014 12:55 PM, Brunner, Brian T. wrote:
Is there any clear documentation describing which conference bridge
slot does what?
Is there any debug setting by which I can tell whether the SplashTone
file is being read and routed to the callers' conference bridge slot?
From my reading the documentation, "somebody" must be putting frames
on the file side, and getting frames on the callers' side, in order
for the audio data to get to the phone.
I don't have a "somebody()" routine, is pjsua/pjmedia (supposedly)
doing this?
There are only a few rtp packets originating from my side.
Media Stats:
[DISCONNCTD] To: "aa2" sip:5@192.168.1.254;tag=be69e5f44p
Call time: 00h:00m:06s, 1st res in 25 ms, conn in 130ms
#0 audio PCMU @8kHz, sendrecv, peer=192.168.1.141:11834
SRTP status: Not active Crypto-suite:
RX pt=0, last update:00h:00m:04.558s ago
total 237pkt 37.9KB (47.4KB +IP hdr) @avg=45.0Kbps/56.3Kbps
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 : 0.000 0.366 3.375 0.125 0.635
TX pt=0, ptime=20, last update:never
total 84pkt 13.4KB (16.8KB +IP hdr) @avg=15.9Kbps/19.9Kbps
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
and
strm0x73d0fc8c .......JB summary:
size=50/eff=50 prefetch=0 level=0
delay (min/max/avg/dev)=0/0/0/0 ms
burst (min/max/avg/dev)=0/0/0/0 frames
lost=0 discard=424 empty=84
I'm guessing this means I never sent a single buffer of audio to the
caller.
I'll try the audio editor method to determine whether the caller sent
voice data.
No data was written to the .wav file
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 31, 2014 5:02 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Some ideas. Using wireshark you can dump the RTP contents of that
stream from .141 and convert it to raw PCMU (using some rtp utils from
columbia, can't remember name) and then import raw pcmu into a sound
editor to play. There may be an easier way too but that's how I've
done it. That could be revealing.
The 7 sec duration is due to a call timer expiring. When call ends the
media stats dump is very useful to see but it isn't in your log, maybe
you need to increase log level.
Your WAV is 48k stereo, I assume 16-bits. Should be fine as pjsip
converts automatically. You might want to try converting it to 16k
mono 16-bit just to be sure the WAV format isn't the issue.
That's all I've got.
Bill
On 10/31/2014 3:58 PM, Brunner, Brian T. wrote:
<quote>Next step: capture wireshark trace and see if media is sent on
RTP?</quote>
Good call: According to WireShark, RTP is between my target (.229)
and the ip-pbx box (.141, which is my gateway to the caller).
The first several RTP packets go both ways, then it's from ip-pbx only.
No. Time Source Destination Protocol Length Info
579 28.662739000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20333, Time=160, Mark
Frame 579: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
580 28.673127000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20334, Time=320
Frame 580: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
581 28.683403000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20335, Time=480
Frame 581: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
582 28.693534000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20336, Time=640
Frame 582: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
583 28.703727000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20337, Time=800
Frame 583: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
584 28.704927000 ip-pbx.bigman.com 192.168.1.229
SIP 610 Request: ACK sip:173@192.168.1.229:5060 |
Frame 584: 610 bytes on wire (4880 bits), 610 bytes captured (4880
bits) on interface 0
Ethernet II, Src: Dell_1e:d1:21 (00:24:e8:1e:d1:21), Dst:
Congatec_0f:f5:6b (00:13:95:0f:f5:6b)
Internet Protocol Version 4, Src: ip-pbx.bigman.com (192.168.1.141),
Dst: 192.168.1.229 (192.168.1.229)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
No. Time Source Destination Protocol Length Info
588 28.718298000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20338, Time=960
Frame 588: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
591 28.731045000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20339, Time=1120
Frame 591: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
592 28.752129000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20340, Time=1280
Frame 592: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
597 28.766290000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20341, Time=1440
Frame 597: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
598 28.771482000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20342, Time=1600
Frame 598: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
599 28.776671000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20343, Time=1760
Frame 599: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
600 28.781771000 192.168.1.229 ip-pbx.bigman.com RTP
214 PT=ITU-T G.711 PCMU, SSRC=0x138B2423, Seq=20344, Time=1920
Frame 600: 214 bytes on wire (1712 bits), 214 bytes captured (1712
bits) on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: terabase (4000), Dst Port: 11432
(11432)
Real-Time Transport Protocol
No. Time Source Destination Protocol Length Info
613 30.407229000 192.168.1.229 ip-pbx.bigman.com UDP
60 Source port: sip Destination port: sip
Frame 613: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
on interface 0
Ethernet II, Src: Congatec_0f:f5:6b (00:13:95:0f:f5:6b), Dst:
Dell_1e:d1:21 (00:24:e8:1e:d1:21)
Internet Protocol Version 4, Src: 192.168.1.229 (192.168.1.229), Dst:
ip-pbx.bigman.com (192.168.1.141)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Data (2 bytes)
NOTE: after 1.74 seconds of bi-directional RTP, (from 28.66 seconds
to 30.40 seconds) RTP from .229 to ip-pbx ceases, and all further RTP
is only from ip-pbx to .229.
RTP messages come from ip-pbx every 20ms except for a 190ms silence
at the end of the 229->141 RTP.
There is one RTCP message from .229 at 30.77
The whole conversation ends 7 seconds after it starts when .229 sends
BYE as it is supposed to.
NOTE: The "PleaseLeaveYourMessage.wav" file is 14MB, lasting 80
seconds. I don't hear any of it.
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 31, 2014 10:10 AM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
The audio device, while in this case is not used to playback or
record, is used to clock media flow. You could alternatively set up
pjsip to use the null-audio device which is just a timer. In your log
I see the audio device is initialized OK:
15:45:40.528 pjsua_aud.c .......Opening sound device PCM@16000/1/20ms
15:45:40.537 pa_dev.c ........Opened device
imx6q-qmx6-sgtl5000: (hw:0,0)(ALSA)/imx6q-qmx6-sgtl5000:
(hw:0,0)(ALSA) for recording and playback, sample rate=16000, ch=1,
bits=16, 320 samples per frame, input latency=100 ms, output
latency=140 ms
And after call connects the player is started:
15:45:40.549 pa_dev.c !Recorder thread started
15:45:40.554 os_core_unix.c Info: possibly re-registering existing
thread
15:45:40.554 pa_dev.c !Player thread started
15:45:40.560 strm0x7570fc8c Start talksprut..
If memory serves, "Start talksprut..." indicates the voice detector
has detected voice onset, so this indicates signal is flowing. It all
looks good.
Next step: capture wireshark trace and see if media is sent on RTP?
Bill
On 10/31/2014 9:01 AM, Brunner, Brian T. wrote:
Logfile: 703 lines/35K attached.
There is no "audio device" involved, as wav file fed from my hard
drive to the Conference Bridge port should be transmitted to the
VoIP phone on the other end.
Which "Initialization" elements do you want to see?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Right, well I asked for the code and I got it, so I'd better
respond. I can see you're answering the call and connecting the
player, so those parts look correct, but it's hard to know if you
are missing something important in initialization. Do you have a
logfile for the auto-answer call? It should show whether the call is
properly connected and may have clues as to why the player isn't
playing. For example, there may be a problem with audio device that
is preventing media flow.
Bill
On 10/30/2014 4:49 PM, Brunner, Brian T. wrote:
Whilst studying the pjproject-2.2.1/pjsip-apps/src/pjsua files for
how to cause a "PleaseLeaveYourMessage auto-play to happen, a few
things:
1: my app (running on ARM7) receives my test call from a company
digital phone via a gateway; this should make no difference.
You asked for example code; after applying what I could grasp
from the auto-play example, here's what I think should work but
doesn't (with lots trimmed out)
if ((status = pjsua_create()) != PJ_SUCCESS) SleepForever("Can't
create PJSUA, is it already running?",status);
if ((status = pjsua_verify_url(LV.MyURI)) != PJ_SUCCESS)
SleepForever("Fix your bad URI, Doofus!",status);
if ((status = pjsua_init(&ep_cfg, &ep_logging_cfg, NULL)) !=
PJ_SUCCESS) SleepForever("Error in pjsua_init()", status);
if ((status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(),
&mod_default_handler)) != PJ_SUCCESS) SleepForever("Can't register
Endpoint", status);
if ((status = pjsua_player_create(&LV.PleaseLeaveYourMessage,
PJMEDIA_FILE_NO_LOOP, &LV.Player_id)) != PJ_SUCCESS)
SleepForever("Can't create SplashPlayer",status);
LV.PlayerConferencePortNo = pjsua_player_get_conf_port(LV.Player_id);
pjsua_player_get_port(LV.Player_id, &LV.SplashPort);
if ((status = pjmedia_wav_player_set_eof_cb(LV.SplashPort, NULL,
&on_playfile_done)) != PJ_SUCCESS) SleepForever("Can't set eof
cb",status);
if ((status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg,
NULL)) != PJ_SUCCESS) SleepForever("Can't create transport, bad
config data?",status);
if ((status = pjsua_acc_add(&acct_cfg, PJ_TRUE, &LV.MyAcctId)) !=
PJ_SUCCESS) SleepForever("Can't create account (you are a
no-account!)",status);
if ((status = pjsua_start()) != PJ_SUCCESS) SleepForever("Can't
start pjsua. Need more chocolate?",status);
The callbacks that appear to matter (with lots trimmed out, you
don't want to read a book):
void cb_on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id
call_id, pjsip_rx_data *rdata)
{
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void cb_on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info CallInfo; pjsua_call_get_info(call_id, &CallInfo);
unsigned MediaIndex; for (MediaIndex = 0; MediaIndex <
CallInfo.media_cnt;MediaIndex++)
{
switch (CallInfo.media[MediaIndex].type)
{
case PJMEDIA_TYPE_AUDIO:
pjsua_conf_connect(LV.PlayerConferencePortNo,
CallInfo.media[MediaIndex].stream.aud.conf_slot);
}
}
}
Then I pick up my phone, dial the gateway, and (when it gives dial
tone) dial this softphone. No "PleaseLeaveYourMessage" plays, but
I can here the softphone hang up on me when the timeout expires.
2: running /usr/bin/pjsip-test-arm-poky-linux-gnueabi -l 6 logs an
error and quits.
line 918060 of 918085.
I can't tell where one test ends and another begins
(hint/beg of benny: please number the tests and print out the
number so I can say "test 4373 failed; it said "foobar" when it
should have said "mumble", and we'd be on the same page together.
Instead:
20:02:58.879 Sending Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) in state Proceeding
20:02:58.879 .Destroying txdata Response msg 100/INVITE/cseq=47769
(tdta0xb93548)
20:02:58.879 .TX 321 bytes Response msg 200/INVITE/cseq=47769
(tdta0xb8c510) to LOOP-DGRAM:0.0.0.0:0:
SIP/2.0 200 OK^M
Via: SIP/2.0/LOOP-DGRAM
129.0.0.1:15060;rport=15060;received=129.0.0.1;branch=z9hG4bK-UAS-Test11^M
Call-ID: 82e44fbf-330c-4350-8d46-773a2226cba8^M
From: sip:alice@127.0.0.1;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: sip:bob@127.0.0.1;tag=z9hG4bK-UAS-Test11^M
CSeq: 47769 INVITE^M
Content-Length: 0^M
^M
--end msg--
20:02:58.879 .State changed from Proceeding to Completed, event=TX_MSG
20:02:58.930 Transport failed to send Response msg
200/INVITE/cseq=47769 (tdta0xb8c510)! Err=120104 (Connection reset
by peer)
20:02:58.932 State changed from Completed to Terminated,
event=TRANSPORT_ERROR
20:02:58.932 error: incorrect status code
20:02:58.932 Timeout timer event
20:02:58.932 .State changed from Terminated to Destroyed, event=TIMER
20:02:58.932 ..Destroying txdata Response msg 200/INVITE/cseq=47769
(tdta0xb8c510)
20:02:58.932 Transaction destroyed!
20:02:58.933 ..ERROR(-170)
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
See pjsua_app.c for example on how to create player and connect to
conference when call answers. It's all part of the --auto-play
option in pjsua.
On 10/27/2014 10:35 AM, Brunner, Brian T. wrote:
I'm puzzled why you need to see broken code in order to tell me
correct code?
Given a call_id, how do I tell which conference port to connect
to, in order to send a wav file to the caller? So far I've
managed to connect it to my sound hardware, which isn't much progress.
I'm curious which call-back is the earliest point in the call
sequence wherein I'm able to tell which conference port to connect
my player to (which I mentioned I'd created).
When we're agreed that my broken code is in the correct place (and
therefore it is the code that is broken, not its location) lets
look at broken code to see what I did wrong in the right place.
Is there an example of this process (I haven't found one yet)?
From: pjsip [pjsip-bounces@lists.pjsip.org] on behalf of Bill
Gardner [billg@wavearts.com]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] How to play a "splash-tone" file as soon as
posssible in the call sequence
Creating a player and connecting to the conference should work,
perhaps you can share some code?
On 10/24/2014 10:50 AM, Brunner, Brian T. wrote:
My attempts (which have been many and unsuccessful) lead me to beg:
How can my SoftVoIPPhone pick up the call and play a "splash
tone" or "please leave your message after the tone" file from the
hard drive to the caller?
I create a player, and I connect it to the conference, but
clearly not the right way or right time.
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