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 at lists.pjsip.org] on behalf of Bill
> Gardner [billg at wavearts.com]
> *Sent:* Thursday, October 30, 2014 5:18 PM
> *To:* pjsip at 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 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 at lists.pjsip.org] on behalf of Bill
>> Gardner [billg at wavearts.com]
>> *Sent:* Monday, October 27, 2014 2:16 PM
>> *To:* pjsip at 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 at lists.pjsip.org] on behalf of Bill
>>> Gardner [billg at wavearts.com]
>>> *Sent:* Friday, October 24, 2014 3:47 PM
>>> *To:* pjsip at 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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>>>
>>>
>>>
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>>
>>
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