Hi,
I use the same soft in 4 devices, and for 2 of them all is ok and the two other bugs every 1-2 days.
The problem is there are no more RTP transmission from my device to the remote system. In the other direction all is ok.
Here is the log of a incomming call :
17:04:21.461 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=102 (rdata0xb5429464)
17:04:21.461 pjsua_core.c .RX 918 bytes Request msg INVITE/cseq=102 (rdata0xb5429464) from UDP 172.27.1.15:5060: INVITE sip:chtel-radiomon-01 at 172.27.36.80:5060;ob SIP/2.0^M Via: SIP/2.0/UDP 172.27.1.15:5060;branch=z
17:04:21.461 pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata0xb5429464)
17:04:21.461 tsx0xb5403d9c ...Transaction created for Request msg INVITE/cseq=102 (rdata0xb5429464)
17:04:21.461 tsx0xb5403d9c ..Incoming Request msg INVITE/cseq=102 (rdata0xb5429464) in state Null
17:04:21.461 tsx0xb5403d9c ...State changed from Null to Trying, event=RX_MSG
17:04:21.462 dlg0xb5419b74 ....Transaction tsx0xb5403d9c state changed to Trying
17:04:21.462 dlg0xb5419b74 ..UAS dialog created
17:04:21.462 dlg0xb5419b74 ..Module mod-invite added as dialog usage, data=0xb5403534
17:04:21.462 dlg0xb5419b74 ...Session count inc to 2 by mod-invite
17:04:21.462 inv0xb5419b74 ..UAS invite session created for dialog dlg0xb5419b74
17:04:21.462 pjsua_media.c ..Call 3: initializing media..
17:04:21.463 pjsua_media.c ...RTP socket reachable at 172.27.36.80:4038
17:04:21.463 pjsua_media.c ...RTCP socket reachable at 172.27.36.80:4039
17:04:21.463 pjsua_media.c ...Media index 0 selected for audio call 3
17:04:21.463 pjsua_call.c ..Call 3: remote NAT type is 0 (Unknown)
17:04:21.463 endpoint ...Response msg 100/INVITE/cseq=102 (tdta0xb5422da8) created
17:04:21.464 dlg0xb5419b74 ...Initial answer Response msg 100/INVITE/cseq=102 (tdta0xb5422da8)
17:04:21.464 inv0xb5419b74 ...Sending Response msg 100/INVITE/cseq=102 (tdta0xb5422da8)
17:04:21.464 dlg0xb5419b74 ....Sending Response msg 100/INVITE/cseq=102 (tdta0xb5422da8)
17:04:21.464 tsx0xb5403d9c ....Sending Response msg 100/INVITE/cseq=102 (tdta0xb5422da8) in state Trying
17:04:21.464 sip_resolve.c .....Target '172.27.1.15:5060' type=UDP resolved to '172.27.1.15:5060' type=UDP (UDP transport)
17:04:21.464 pjsua_core.c .....TX 288 bytes Response msg 100/INVITE/cseq=102 (tdta0xb5422da8) to UDP 172.27.1.15:5060: SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 172.27.1.15:5060;received=172.27.1.15;branch=z9hG4bK0a3d8e
17:04:21.464 tsx0xb5403d9c .....State changed from Trying to Proceeding, event=TX_MSG
17:04:21.464 dlg0xb5419b74 ......Transaction tsx0xb5403d9c state changed to Proceeding
New call with id [3]
17:04:21.464 pjsua_call.c ..Answering call 3: code=200
17:04:21.464 inv0xb5419b74 ....SDP negotiation done, status=0
17:04:21.465 pjsua_call.c .....Call 3: remote NAT type is 0 (Unknown)
17:04:21.465 pjsua_media.c .....Call 3: updating media..
17:04:21.465 pjsua_aud.c ......Audio channel update..
17:04:21.465 strm0xb5431044 .......VAD temporarily disabled
17:04:21.465 rtp.c .......pjmedia_rtp_session_init: ses=0xb5432b08, default_pt=9, ssrc=0x5d980063
17:04:21.465 rtp.c .......pjmedia_rtp_session_init: ses=0xb5433190, default_pt=9, ssrc=0x5d980063
17:04:21.465 stream.c .......Stream strm0xb5431044 created
17:04:21.465 strm0xb5431044 .......Encoder stream started
17:04:21.465 strm0xb5431044 .......Decoder stream started
17:04:21.465 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv)
Play sound file [/usr/share/sounds/chatel1.wav] for id [2]
Lock - playSoundDuringCall - try
Trying to get pj lock...
Lock - playSoundDuringCall - ok
17:04:21.466 pjsua_aud.c .....Creating file player: /usr/share/sounds/chatel1.wav..
17:04:21.466 wav_player.c ......File player '/usr/share/sounds/chatel1.wav' created: samp.rate=8000, ch=1, bufsize=4KB, filesize=19KB
17:04:21.466 resample.c ......resample created: high qualiy, large filter, in/out rate=8000/16000
17:04:21.466 resample.c ......resample created: high qualiy, large filter, in/out rate=16000/8000
17:04:21.466 pjsua_aud.c ......Player created, id=6, slot=9
Unlock - playSoundDuringCall - try
Unlock - playSoundDuringCall - ok
17:04:21.466 pjsua_aud.c .....Conf connect: 9 --> 1
17:04:21.466 conference.c ......Port 9 (/usr/share/sounds/chatel1.wav) transmitting to port 1 (sip:2901 at 172.27.1.15)
Play sound file [/usr/share/sounds/chatel1.wav] for id [3]
Lock - playSoundDuringCall - try
Trying to get pj lock...
Lock - playSoundDuringCall - ok
17:04:21.467 pjsua_aud.c .....Creating file player: /usr/share/sounds/chatel1.wav..
17:04:21.467 wav_player.c ......File player '/usr/share/sounds/chatel1.wav' created: samp.rate=8000, ch=1, bufsize=4KB, filesize=19KB
17:04:21.467 resample.c ......resample created: high qualiy, large filter, in/out rate=8000/16000
17:04:21.467 resample.c ......resample created: high qualiy, large filter, in/out rate=16000/8000
17:04:21.467 pjsua_aud.c ......Player created, id=9, slot=12
Unlock - playSoundDuringCall - try
Unlock - playSoundDuringCall - ok
17:04:21.467 pjsua_aud.c .....Conf connect: 12 --> 7
17:04:21.467 conference.c ......Port 12 (/usr/share/sounds/chatel1.wav) transmitting to port 7 (sip:003 at 172.27.1.15)
17:04:21.467 pjsua_aud.c .....Conf connect: 0 --> 7
17:04:21.467 conference.c ......Port 0 (USB PnP Sound Device: USB Audio (hw:1,0) (44KHz)) transmitting to port 7 (sip:003 at 172.27.1.15)
17:04:21.467 inv0xb5419b74 ....Sending Response msg 200/INVITE/cseq=102 (tdta0xb5422da8)
17:04:21.467 dlg0xb5419b74 .....Sending Response msg 200/INVITE/cseq=102 (tdta0xb5422da8)
17:04:21.467 tsx0xb5403d9c .....Sending Response msg 200/INVITE/cseq=102 (tdta0xb5422da8) in state Proceeding
17:04:21.468 pjsua_core.c ......TX 834 bytes Response msg 200/INVITE/cseq=102 (tdta0xb5422da8) to UDP 172.27.1.15:5060: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 172.27.1.15:5060;received=172.27.1.15;branch=z9hG4bK0a3d8e1c^M
17:04:21.468 tsx0xb5403d9c ......State changed from Proceeding to Completed, event=TX_MSG
17:04:21.468 dlg0xb5419b74 .......Transaction tsx0xb5403d9c state changed to Completed
Sound message has been play recently, don't play this one.
17:04:21.487 sip_endpoint.c Processing incoming message: Request msg ACK/cseq=102 (rdata0xb5429464)
17:04:21.487 pjsua_core.c .RX 437 bytes Request msg ACK/cseq=102 (rdata0xb5429464) from UDP 172.27.1.15:5060: ACK sip:chtel-radiomon-01 at 172.27.36.80:5060;ob SIP/2.0^M Via: SIP/2.0/UDP 172.27.1.15:5060;branch=z9hG4bK
I see than the conference bridge are increasing value (when all is ok the value donesn't increase). But i can't understand why.
Here is the callback call on mediaState :
static void onCallMediaState(pjsua_call_id callId)
{
pjsua_call_info callInfo;
unsigned mi;
pj_bool_t hasError = PJ_FALSE;
pjsua_call_get_info(callId, &callInfo);
for (mi = 0; mi < callInfo.media_cnt; ++mi)
{
switch (callInfo.media[mi].type)
{
case PJMEDIA_TYPE_AUDIO:
onCallAudioState(&callInfo, mi, &hasError);
break;
default:
break;
}
}
}
static void onCallAudioState(pjsua_call_info *callInfo, unsigned mediaInfo,
pj_bool_t *has_error)
{
PJ_UNUSED_ARG(has_error);
pjsua_conf_port_id callConfSlot;
callConfSlot = callInfo->media[mediaInfo].stream.aud.conf_slot;
if (callInfo->media[mediaInfo].status == PJSUA_CALL_MEDIA_ACTIVE
|| callInfo->media[mediaInfo].status == PJSUA_CALL_MEDIA_REMOTE_HOLD)
{
// Put call in conference with other calls, if desired
pjsua_call_id callIds[PJSUA_MAX_CALLS];
unsigned callCnt = PJ_ARRAY_SIZE(callIds);
unsigned i;
/*
* Get all calls, and establish media connection between
* this call and other calls.
*/
pjsua_enum_calls(callIds, &callCnt);
for (i = 0; i < callCnt; ++i)
{
if (!pjsua_call_has_media(callIds[i]))
{
continue;
}
if (callIds[i] == callInfo->id)
{
continue;
}
char* answerPath = cfg_getstr(systemConfiguration, SOUND_ANSWER);
playSoundFromConfiguration(answerPath, callIds[i]);
}
if (pjsua_conf_connect(0, callConfSlot) != PJ_SUCCESS)
{
syslog(LOG_INFO, "Fail to connect to the conference port");
}
}
}
I have see I forget to set has_error flag on audio call I have add this part to my new version, but I think this isn't the problem.
?
Thanks
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