Closed VincentGijsen closed 9 years ago
Hi,
first-off, nice work on the wrapper!
I've got it almost working, but the audio is ginving me troubles, not sure how to fix this. my setup: laptop(pjsip4net) ->asterisk (linux) -> sip-device
I can register to asterisk, and dial, however I hear for 0,5 sec crippled audio
my app.config:
<sipua srtp="Disabled" secureSignaling="0" logMessages="true" logLevel="5" traceAndDebug="false" autoAnswer="true" autoRecording="false" autoConference="false" maxCalls="1" > <accounts> <!--<account accountId="sip:1002@192.168.10.254:5080" registrarUri="sip:192.168.10.1:5060" userName="1002" password="1002" realm="*" isDefault="true" publishPresence="false" /> --> </accounts> <sipTransport type="udp" Port="5080" /> <media isVADEnabled="true" Quality="5" /> </sipua>
I set my codec to PCMU/8000/1, prio 1
and dail my sipSpeaker:
call = ua.CallManager.MakeCall(x => x.To("1000") .At("192.168.10.1") .From(ua.AccountManager.GetAccountById(Convert.ToInt32(1))) .Call() );
and attach it to my conference:
ua.MediaManager.ConferenceBridge.ConnectCall(call); Debug.WriteLine("cap: "+ ua.MediaManager.CurrentCaptureDevice.Name); Debug.WriteLine("play: " + ua.MediaManager.CurrentPlaybackDevice.Name);
my log:
2015-07-28 15:14:11,461 [13] DEBUG - 15:14:11.461 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,481 [13] DEBUG - 15:14:11.481 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,649 [13] DEBUG - 15:14:11.649 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,650 [13] DEBUG - 15:14:11.650 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,650 [13] DEBUG - 15:14:11.650 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,651 [13] DEBUG - 15:14:11.651 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:11,651 [13] DEBUG - 15:14:11.651 ec0A962F88 Underflow, buf_cnt=1, will generate 1 frame 2015-07-28 15:14:12,446 [13] DEBUG - 15:14:12.446 sound_port.c EC suspended because of inactivity
Any idea's how to tackle this last hurdle?
It seems to me that the buffer size is too small, I don't care for more latency
cheers, Vincent
Mmm very strange, whenever I use GSM or G722 as codecs, it works as expected...
Hi,
first-off, nice work on the wrapper!
I've got it almost working, but the audio is ginving me troubles, not sure how to fix this. my setup: laptop(pjsip4net) ->asterisk (linux) -> sip-device
I can register to asterisk, and dial, however I hear for 0,5 sec crippled audio
my app.config:
I set my codec to PCMU/8000/1, prio 1
and dail my sipSpeaker:
and attach it to my conference:
my log:
Any idea's how to tackle this last hurdle?
It seems to me that the buffer size is too small, I don't care for more latency
cheers, Vincent