Hi......
Does any one has the idea on the following requirement.Your guidance is
highly appreciated...
On Mon, Oct 15, 2012 at 2:21 PM, john smith <pyroflares at gmail.com> wrote:
> Hi Folks,
>
> I am trying to create a conference enviorment for the following requirement
>
> 1.)accepting calls from n no of b-party in such a manner that the voice
> from b-party is heard to all the users in the conference and vice versa.
> 2.)making calls from a-party to n-users in such a manner that the voice
> from a-party is heard to all the users in the conference and vice versa.
>
> i have gone through confsample.c to get an idea.However i am successful in
> registering my account to my sip server but when i try to do
> pjsip_conf_add_port
> in my code i get the following error....
>
>
> 14:23:29.785 pjsua_media.c Media index 0 selected for call 0
> 14:23:29.785 pjsua_core.c TX 481 bytes Response msg 100/INVITE/cseq=1
> (tdta0x212a5f0) to UDP 172.22.0.15:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.22.0.15:5060
> ;received=172.22.0.15;branch=z9hG4bKd9ea.2595dbb.1
> Via: SIP/2.0/UDP 172.22.25.245:5086
> ;rport=5086;branch=z9hG4bK462402ec-1215-e211-87fa-00137237e9b1
> Record-Route: <sip:172.22.0.15;lr;did=32f.ff82e9b>
> Call-ID: 20f600ec-1215-e211-87fa-00137237e9b1 at localhost.localdomain
> From: "john.smith" <sip:ekigatest@172.22.0.15
> >;tag=2efc00ec-1215-e211-87fa-00137237e9b1
> To: <sip:2315 at 172.22.0.15>
> CSeq: 1 INVITE
> Content-Length: 0
>
>
> --end msg--
> 14:23:29.785 APP Incoming call from "john.smith" <
> sip:ekigatest at 172.22.0.15>!!
> 14:23:29.785 strm0x212fd48 VAD temporarily disabled
> 14:23:29.786 strm0x212fd48 Encoder stream started
> 14:23:29.786 strm0x212fd48 Decoder stream started
> 14:23:29.786 pjsua_media.c Media updates, stream #0: speex (sendrecv)
> 14:23:29.786 pjlib select() I/O Queue created (0x21426e8)
> 14:23:29.786 pjsua_media.c Opening sound device PCM at 16000/1/20ms
>
> &&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
> 14:23:29.834 ec0x214eef0 AEC created, clock_rate=16000, channel=1,
> samples per frame=320, tail length=200 ms, latency=100 ms
> 14:23:29.835 conference.c Port 1 (sip:ekigatest at 172.22.0.15)
> transmitting to port 0 (Yamaha DS-1 (YMF724F): YMFPCI (hw:0,0))
> 14:23:29.835 conference.c Port 0 (Yamaha DS-1 (YMF724F): YMFPCI
> (hw:0,0)) transmitting to port 1 (sip:ekigatest at 172.22.0.15)
> 14:23:29.872 os_core_unix.c Info: possibly re-registering existing thread
> 14:23:30.491 strm0x212fd48 VAD re-enabled
> Expression 'ret' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1026
> Expression 'AlsaOpen( &alsaApi->baseHostApiRep, params, streamDir,
> &self->pcm )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1183
> Expression 'PaAlsaStreamComponent_Initialize( &self->capture, alsaApi,
> inParams, StreamDirection_In, NULL != callback )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1410
> Expression 'PaAlsaStream_Initialize( stream, alsaHostApi, inputParameters,
> outputParameters, sampleRate, framesPerBuffer, callback, streamFlags,
> userData )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1991
> Expression 'ret' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1026
> Expression 'AlsaOpen( &alsaApi->baseHostApiRep, params, streamDir,
> &self->pcm )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1183
> Expression 'PaAlsaStreamComponent_Initialize( &self->capture, alsaApi,
> inParams, StreamDirection_In, NULL != callback )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1410
> Expression 'PaAlsaStream_Initialize( stream, alsaHostApi, inputParameters,
> outputParameters, sampleRate, framesPerBuffer, callback, streamFlags,
> userData )' failed in
> 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1991
> 14:23:32.248 *APP Unable to create conference bridge: Device
> unavailable [code=469984]*
> Segmentation fault
>
> Heres my code...which is called from main.....Please guide me on this
> .................................
>
>
>
> static void on_call_media_state(pjsua_call_id call_id)
> {
> pj_caching_pool cp;
> pjmedia_endpt *med_endpt;
> pj_pool_t *pool;
> pjmedia_conf *conf;
> pj_status_t status;
>
> int port_count;
> pjsua_call_info ci;
>
> pjsua_call_get_info(call_id, &ci);
>
>
> pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0);
>
> /*
> * Initialize media endpoint.
> * This will implicitly initialize PJMEDIA too.
> */
> status = pjmedia_endpt_create(&cp.factory, NULL, 1, &med_endpt);
> //PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
>
> /* Create memory pool to allocate memory */
> pool = pj_pool_create( &cp.factory, /* pool factory */
> "wav", /* pool name. */
> 4000, /* init size */
> 4000, /* increment size */
> NULL /* callback on error */
> );
>
>
> if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
> // When media is active, connect call to sound device.
> pjsua_conf_connect(ci.conf_slot, 0);
> pjsua_conf_connect(0, ci.conf_slot);
> }
>
>
> status = pjmedia_conf_create( pool, /* pool to use */
> 1,/* number of ports */
> 44100,
> 1,
> 882,
> 16,
> 0, /* options */
> &conf /* result */
> );
> if (status != PJ_SUCCESS)
> app_perror(THIS_FILE, "Unable to create conference bridge",
> status);
> // printf("app_perror(THIS_FILE, Unable to create conference
> bridge\n");
>
>
>
> status =
> pjmedia_conf_add_port(conf,pool,(pjmedia_port*)ci.conf_slot,NULL,NULL);
> if (status != PJ_SUCCESS) {
> app_perror(THIS_FILE, "Unable to add conference port", status);
> printf("Unable to add conference port add port to bridge\n");
> //return 1;
> }
>
> }
>
>
> Thanks & Regards
> John Smith
>
>
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