How to get the audio stream from PJSIP when there is no audio hardware device

Question

I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device .

I'm unsure about the details, but the sparse documentation for PJSIP suggests it should be possible through the pjsua_set_null_snd_dev() call.

In the fully functioning (Windows biased) example below the call pjmedia_aud_dev_default_param(PJMEDIA_AUD_DEFAULT_CAPTURE_DEV, &param) returns PJMEDIA_AUD_INVALID_DEV in the status.

The code generates this same error on Linux (Ubuntu 14) and Windows 10 when there are no hardware audio devices present. If there is an hardware audio device driver installed the exact same code works fine on both OSes.

I have compiled the PJSIP libraries with PJMEDIA_AUDIO_DEV_HAS_NULL_AUDIO enabled. On Linux the presence of the module snd-dummy does not help.

How do I get access to the audio data stream from a SIP call after calling pjsua_set_null_snd_dev()?

#include <pjlib.h>
#include <pjlib-util.h>
#include <pjnath.h>
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjsip_simple.h>
#include <pjsua-lib/pjsua.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pj/log.h>
#include <pj/os.h>

    int main(int, char **)
    {
    // Create pjsua first! 
            pj_status_t status = pjsua_create();
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr,"pjsua_create error\n");
                return -1;
            }

    // Init pjsua 
            pjsua_config cfg;
            pjsua_logging_config log_cfg;

            pjsua_config_default(&cfg);

            pjsua_logging_config_default(&log_cfg);
            log_cfg.console_level = 4;

            status = pjsua_init(&cfg, &log_cfg, NULL);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr,"pjsua_init error\n");
                return -1;
            }


    // Proactively list known audio devices so we are sure there are NONE
            pjmedia_aud_dev_info info[64];
            unsigned info_count = 64;
            pjsua_enum_aud_devs(info, &info_count);

            fprintf(stderr,"Listing known sound devices, total of [%u]\n", info_count);
            for (unsigned i = 0; i<info_count; ++i)
            {
                fprintf(stderr,"Name [%s]", info[i].name);
            }

    // Add transport
            pjsua_transport_config tcfg;
            pjsua_transport_id trans_id;
            pjsua_transport_config_default(&tcfg);
            tcfg.port = 5060;
            status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &tcfg, &trans_id);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "pjsua_transport_create error\n");
                return -1;
            }

    // Initialization is done, now start pjsua 
            status = pjsua_start();
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "pjsua_start error\n");
                return -1;
            }

    // Set NULL sound
            status = pjsua_set_null_snd_dev();
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "pjsua_set_null_snd_dev error");
                return -1;
            }

    // Register to a SIP server by creating SIP account, I happen use use Asterisk 
            pjsua_acc_id acc_id;
            fprintf(stderr, "Setting up SIP server registration\n");
            {
                pjsua_acc_config cfg;
                pjsua_acc_config_default(&cfg);
                cfg.id = pj_str("sip:[email protected]");
                cfg.reg_uri = cfg.id; // same as ID
                cfg.cred_count = 1;

                cfg.cred_info[0].realm = pj_str("*");
                cfg.cred_info[0].scheme = pj_str("digest"); 
                cfg.cred_info[0].username = pj_str("6001");
                cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
                cfg.cred_info[0].data = pj_str("teddy");

                status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
                if (status != PJ_SUCCESS)
                {
                    fprintf(stderr, "pjsua_acc_add error\n");
                    return -1;
                }
            }

            fprintf(stderr, "Waiting for SIP server registration to complete....\n");

            Sleep(2000); // sleep 2 seconds

    // Call extension 9 on my Asterisk server at 10.0.0.21:5060
            pj_str_t sip_target(pj_str("sip:[email protected]"));
            fprintf(stderr, "Making call to [%s]\n", sip_target.ptr);

            pjsua_call_id call_id;
            status = pjsua_call_make_call(acc_id, &sip_target, 0, NULL, NULL, &call_id);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "pjsua_call_make_call error\n");
                return -1;
            }

            pj_pool_t * pool = nullptr;
            pjmedia_port * wav = nullptr;
            pjmedia_aud_stream *strm = nullptr;
            pool = pj_pool_create(pjmedia_aud_subsys_get_pool_factory(), "wav-audio", 1000, 1000, NULL);

            if (nullptr == pool)
            {
                fprintf(stderr,"Pool creation failed\n");
                return -1;
            }

    // 8kHz, single channel 16bit MS WAV format file
            status = pjmedia_wav_writer_port_create(pool, "test.wav", 8000, 1, 320, 16, PJMEDIA_FILE_WRITE_PCM, 0, &wav);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "Error creating WAV file\n");
                return -1;
            }

            pjmedia_aud_param param; 
    //////////////////////////////////////////////////////
    // FAILURE HERE : This is the function call which returns PJMEDIA_AUD_INVALID_DEV
    //////////////////////////////////////////////////////
            status = pjmedia_aud_dev_default_param(PJMEDIA_AUD_DEFAULT_CAPTURE_DEV, &param); 
            if (status != PJ_SUCCESS) 
            {
                fprintf(stderr, "pjmedia_aud_dev_default_param()");
                return -1;
            }

            param.dir = PJMEDIA_DIR_CAPTURE;
            param.clock_rate = PJMEDIA_PIA_SRATE(&wav->info);
            param.samples_per_frame = PJMEDIA_PIA_SPF(&wav->info);
            param.channel_count = PJMEDIA_PIA_CCNT(&wav->info);
            param.bits_per_sample = PJMEDIA_PIA_BITS(&wav->info);

            status = pjmedia_aud_stream_create(&param, &test_rec_cb, &test_play_cb, wav, &strm);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "Error opening the sound stream");
                return -1;
            }

            status = pjmedia_aud_stream_start(strm);
            if (status != PJ_SUCCESS)
            {
                fprintf(stderr, "Error starting the sound device");
                return -1;
            }

    // Spend some time allowing the called party to pick up and recording to proceed
            Sleep(10000); // sleep 10 seconds

    // Clean up code omitted
    return 0;
    }

Apologies to the pure of heart for the mix of C and C++ above.

Answer 1

Solved this by loading the Alsa module snd-dummy.

Look in /lib/modules/YOUR_KERNEL_VERSION/modules.dep if its mentioned.

If you have it then load it with modprobe snd-dummy

Otherwise recompile your Kernel to include it as a module or follow the installation in the link above.