The big bug is:
When pjsip usa works with asterisk, After it's put on hold, listening to on hold music, or transfered , listening to hold music, It's very possible to be deaf. Only after you send reinvite, rebuild the media channel, then resume.
So this is a big bug, I hope somebody can solve it.
I solve it by changing sound_prot.c, after a little time(1 second), the media port can not get any sound fram, I make it send reinvite. Look the codes below:
in function play_cb
return PJ_SUCCESS;
no_frame:
if (pjsua_call_get_count() > 0)
{
if (pjsua_call_has_media(0)){
++snd_port->no_frame_count;
if (snd_port->no_frame_count > snd_port->ec_suspend_limit/AEC_SUSPEND_LIMIT*1.1)
{
pjsua_call_reinvite(0, PJ_TRUE, NULL);
snd_port->no_frame_count = 0;
}
}
}
if (snd_port->ec_state && !snd_port->ec_suspended) {
++snd_port->ec_suspend_count;
if (snd_port->ec_suspend_count > snd_port->ec_suspend_limit) {
snd_port->ec_suspended = PJ_TRUE;
PJ_LOG(4,(THIS_FILE, "EC suspended because of inactivity"));
}
if (snd_port->ec_state) {
/* To maintain correct delay in EC */
pjmedia_echo_playback(snd_port->ec_state, (pj_int16_t*)output);
}
}
The Blue part is the change. I and a new variable no_frame_count in sn_port,It caculate how many frames not received. If more than 1.1 seconds frames lost, I send reinvite.
I know this is not the final solution, But it works.
I hope somebody can solve it in a final righ way.
Regards
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