Re: Please help - wav_writer writing silence when switch to 1 conference bridge PER call (following pjsua-lib-perf FAQ)

Michael Leonard <emailmleonard@xxxxxxxxx> · Tue, 4 Oct 2016 14:59:31 +0100

Hi Bill. Thanks for the quick response.
I'm not sure if you saw it: my entire code is in one file called whole_app.c which is included in the gist (scroll to the bottom to see the file: https://gist.github.com/MikeLeonard/6b8b4986d6529ae751ebd2af5bf4d29f)

I'm using pjmedia_conf_add_port inside of on_stream_created, which (I think) adds the stream port (p_port) to the conference bridge (cd->conf) and saves the slot number into &cd->call_slot.

static void on_stream_created(pjsua_call_id call_id, 
                              pjmedia_stream *strm,
                              unsigned stream_idx, 
                              pjmedia_port **p_port){
	log_message("RUNNING... on_stream_created\n");						  
	struct call_data *cd;
	cd = (struct call_data*) pjsua_call_get_user_data(call_id);
	if (!cd)
		return;
	pjmedia_conf_add_port(cd->conf, cd->pool, *p_port, NULL, &cd->call_slot);
}

Elsewhere in call_media_init which is called when the call state is 'CALLING', I'm running the following:
pjmedia_conf_connect_port(cd->conf, cd->call_slot, cd->writer_slot, 0); //flow audio from call to writer
pjmedia_conf_connect_port(cd->conf, cd->player_slot, cd->call_slot, 0); //flow wav reader audio to call
... to connect the call_slot to the writer_slot and the player_slot.

Do I need to make another connection? Perhaps to connect the master port and/or the null port to one or more of these? I'm afraid that I haven't been able to really get to grips with these two ports, I don't really understand what they are for. (I've just read that I need them when there's no sound device in order to drive the get_frame and put_frame callbacks - plus the FAQ uses them).

Thanks again for the help

Mike

.............................................................................................................................................................................
Also, I've just realised a potential problem: on_stream_created is called AFTER the call state is 'CALLING' (which calls my call_media_init function)... so I guess that call_media_init is trying to connect things to call_slot BEFORE call_slot is set by pjmedia_conf_add_port inside of on_stream_created. I couldn't really tell from the FAQ exactly where I should put these functions - do you know where I've gone wrong?

UPDATE: Nope. I've just tried changing it so that call_media_init is called when the MEDIA state is 'ACTIVE' instead of when the call state is CALLING, so that call_media_init is called after on_stream_created  (which calls pjmedia_conf_add_port(cd->conf, cd->pool, *p_port, NULL, &cd->call_slot)). However I get the same silence. 

Anyway, the cod has some print commands which show the order things run in, in case that's helpful to you:

/myapp sip:+447XXXXXXXX@xxxxxxxxxxxxxxxxxxxxxx >> log.txt
WARNING: no real random source present!
RUNNING... on_call_state
Call state is CALLING.
RUNNING... call_media_init      
RUNNING... on_call_state
RUNNING... on_stream_created       
RUNNING... on_call_media_state
Media state is ACTIVE.
RUNNING... on_call_state
RUNNING... on_call_state
RUNNING... on_call_state
Call state is CONFIRMED.
RUNNING... on_call_state
Call state is DISCONNECTED.
RUNNING... call_media_deinit
Called call_media_deinit.
RUNNING... on_stream_destroyed
^C

Thanks

Mike

On 4 October 2016 at 14:02, Bill Gardner <billg@xxxxxxxxxxxx> wrote:

    Hi Mike,

    I don't see any code that connects the stream port (codec, jitter,
    RTP, etc) to your conference bridge at call_slot. Is this done
    elsewhere?

    Regards,

    Bill

    On 10/4/2016 7:30 AM, Michael Leonard
      wrote:

      Hello everyone

        Please help! My wav_writer is just writing silence (of the
          correct length) when trying to use pjsua-lib with 1 conference
          bridge per call, as explained in the FAQ here: https://trac.pjsip.org/repos/wiki/FAQ#pjsua-lib-perf.
          Note that when I use the default pjsua conference bridge
          everything works fine for me.

        I've saved the code and a log file to a github public gist
          here: https://gist.github.com/MikeLeonard/6b8b4986d6529ae751ebd2af5bf4d29f

        I would *really* appreciate it if someone could look
          through call_media_init function I've written (also pasted
          below) and tell me what I'm doing wrong. 

        If it takes someone a lot of time I'd be happy to buy them
          a coffee (or amazon voucher!) to say thank you.

        Thanks very much in advance

        Mike

        More information:

        ..................................................................................................................................................
        **I think I've ruled out problems with my setup
        - I'm trying to create a server that makes calls to a
          regular mobile or landline from a server (ie no voip phone so
          no sound device), and records the speech of the receiver to a
          wav file.

          I'm running this on ubuntu, inside a docker container, on
          a macbook, and am making an outbound call to a mobile number
          via a twilio sip trunk. I'm also compiling this using C++
          (because I need various C++ libraries for what I have in mind
          once I get the basics working). 

        The above setup ALL works fine - I modified
          samples/simple_pjsua.c and got a wav_writer and a wav_player
          working no problem with the default pjsua conference bridge.
          So I'm pretty sure the problem is with how I've tried to
          follow the pjsua-lib-perf FAQ. 

        **I can't see any issues in the log file, and I've
            confirmed the key 'on_' callbacks all definitely run

          - The log file prints out the following - which to me
            looks like I've connected the right ports, but I don't
            really understand this to be honest I've just be copying
            examples: 
          09:59:05.565            APP  .......Call 0 state=CALLING
          09:59:05.572   wav_writer.c  .......File writer
            'testingtesting.wav' created: samp.rate=16000, bufsize=4KB
          09:59:05.572   conference.c  .......Port 0 (Master/sound)
            transmitting to port 1 (testingtesting.wav)
          09:59:05.579   wav_player.c  .......File player
            'message.wav' created: samp.rate=16000, ch=1, bufsize=4KB,
            filesize=56KB
          09:59:05.579   conference.c  .......Port 2 (message.wav)
            transmitting to port 0 (Master/sound) 

            - I'm just trying one call at the moment - I haven't
              tried multiple calls yet.

            - I've confirmed that the key callbacks all run: on_call_state, on_call_media_state, on_stream_created and on_stream_destroyed

          - I've added lots of "if (status != PJ_SUCCESS)
            {pjsua_perror(THIS_FILE, "STATUS ERROR: ", status);}" into
            the code, but don't get any "STATUS ERROR" messages in the
            log.

        **A wav file is indeed written

          - ...it just contains silence. It seems to be silence of
            the length of the call however. 

          I've also tried looping back the audio, and playing a wav
            file to the receiver, but I just get silence. These all work
            in my basic samples/simple_pjsua.c version with the default
            pjsua conference bridge.

        ..................................................................................................................................................

        Key piece of code... see github public gist for full
            code + log file https://gist.github.com/MikeLeonard/6b8b4986d6529ae751ebd2af5bf4d29f

          struct call_data {
            pj_pool_t           *pool;
            pjmedia_conf        *conf;
            pjmedia_port        *cport;
            pjmedia_port        *null;
            pjmedia_port        *writer;
            pjmedia_port        *player;
            pjmedia_master_port *m;
            unsigned int                  call_slot;
            unsigned int                  writer_slot;
            unsigned int                  player_slot;
          };

          static void call_media_init(pjsua_call_id call_id){
          	log_message("RUNNING...
            call_media_init\n");

          	pj_pool_t
            *pool;
          	struct
            call_data *cd;
          	pj_status_t
            status;

          	pool
            = pjsua_pool_create("mycall", 4000, 4000);
          	cd
            = PJ_POOL_ZALLOC_T(pool, struct call_data);
          	cd->pool
            = pool;

          	pjsua_call_set_user_data(call_id,
            (void*)cd);

          	pjsua_media_config
            media_cfg;
          	pjsua_media_config_default(&media_cfg);

          	status
            = pjmedia_conf_create(
          		cd->pool, 
          		media_cfg.max_media_ports,
             //max media ports
          		media_cfg.clock_rate,
          		media_cfg.channel_count,
          		media_cfg.clock_rate
            * media_cfg.channel_count * media_cfg.audio_frame_ptime /
            1000, //mconf_cfg.samples_per_frame,
          		16,
            //mconf_cfg.bits_per_sample,
          		PJMEDIA_CONF_NO_DEVICE
            | PJMEDIA_CONF_NO_MIC,  //options
          		&cd->conf
            //pointer to conference bridge instance
          	);
          	if
            (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE, "STATUS
            ERROR: ", status);}
          	cd->cport
            = pjmedia_conf_get_master_port(cd->conf);

          	status
            = pjmedia_null_port_create(
          		cd->pool,
          		media_cfg.clock_rate,
          		media_cfg.channel_count,
          		media_cfg.clock_rate
            * media_cfg.channel_count * media_cfg.audio_frame_ptime /
            1000, //mconf_cfg.samples_per_frame,
          		16,
            //mconf_cfg.bits_per_sample,
          		&cd->null);
          	if
            (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE, "STATUS
            ERROR: ", status);}
          	status
            = pjmedia_master_port_create(cd->pool, cd->null,
            cd->cport, 0, &cd->m);
          	if
            (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE, "STATUS
            ERROR: ", status);}
          	status
            = pjmedia_master_port_start(cd->m);
          	if
            (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE, "STATUS
            ERROR: ", status);}
             //todo(mike) handle errors, see pjsua_aud.c

             /* wav writer */
          	status
            = pjmedia_wav_writer_port_create(
          		cd->pool, 
          		"testingtesting.wav",
            //path
          		media_cfg.clock_rate,
          		media_cfg.channel_count,
          		media_cfg.clock_rate
            * media_cfg.channel_count * media_cfg.audio_frame_ptime /
            1000, //mconf_cfg.samples_per_frame,
          		16,
            //mconf_cfg.bits_per_sample,
          		0,
            //options
          		0,
            //buf_size defaults to 4kb if set to 0
          		&cd->writer
            //yes this should be a pjmedia_port **
          		); 
          	if
            (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE, "STATUS
            ERROR: ", status);}
          	status
            = pjmedia_conf_add_port(cd->conf, cd->pool,
            cd->writer, NULL, &cd->writer_slot);
          	if
            (status != PJ_SUCCESS) {
          		pjsua_perror(THIS_FILE,
            "STATUS ERROR: ", status);
          		pjmedia_port_destroy(cd->writer);
          	}

          	pjmedia_conf_connect_port(cd->conf,
            cd->call_slot, cd->writer_slot, 0);

          	/*
            wav player */
          	status
            = pjmedia_wav_player_port_create(
          		cd->pool, 
          		"message.wav",
          		media_cfg.audio_frame_ptime,
          		0, 
          		0, 
          		&cd->player
             //yes this should be a pjmedia_port **
          		);
              if (status != PJ_SUCCESS) {pjsua_perror(THIS_FILE,
            "STATUS ERROR: ", status);}

          	status
            = pjmedia_conf_add_port(cd->conf, cd->pool,
            cd->player, NULL, &cd->player_slot);
          	if
            (status != PJ_SUCCESS) {
          		pjsua_perror(THIS_FILE,
            "STATUS ERROR: ", status);
          		pjmedia_port_destroy(cd->player);
          	}

          	pjmedia_conf_connect_port(cd->conf,
            cd->player_slot, cd->call_slot, 0);

          	//uncomment
            to loop back remote audio (also doesn't work)
          	//pjmedia_conf_connect_port(cd->conf,
            cd->call_slot, cd->call_slot, 0);
          }

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