<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 at 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@xxxxxxxxxxxxxxx] on behalf of Bill Gardner [billg at wavearts.com]
Sent: Friday, October 31, 2014 10:10 AM
To: pjsip at lists.pjsip.org
Subject: Re: 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 at 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@xxxxxxxxxxxxxxx<mailto:pjsip-bounces at lists.pjsip.org>] on behalf of Bill Gardner [billg at wavearts.com<mailto:billg at wavearts.com>]
Sent: Thursday, October 30, 2014 5:18 PM
To: pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
Subject: Re: 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><sip:alice at 127.0.0.1>;tag=d71eff2a-a70f-4e7c-9151-a3a8a2f5e2f4^M
To: <sip:bob at 127.0.0.1><sip:bob at 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@xxxxxxxxxxxxxxx<mailto:pjsip-bounces at lists.pjsip.org>] on behalf of Bill Gardner [billg at wavearts.com<mailto:billg at wavearts.com>]
Sent: Monday, October 27, 2014 2:16 PM
To: pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
Subject: Re: 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@xxxxxxxxxxxxxxx<mailto:pjsip-bounces at lists.pjsip.org>] on behalf of Bill Gardner [billg at wavearts.com<mailto:billg at wavearts.com>]
Sent: Friday, October 24, 2014 3:47 PM
To: pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
Subject: Re: 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 at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20141031/39da179e/attachment.html>
