Hi,
here is a patch for the DTMF problem.
I am still trying to figure out how to best solve the supervised transfer
problem.
...Carsten
Carsten Avenhaus wrote:
Paul Whitfield wrote:
For my application I have re-worked the code to not call
MprDecode::pushPacket() for DTMF packets,
this then caused a problem that the DTMF packets
never got decoded, I fixed this and my application is working fine now.
I thought about that too, but that means DTMFs can probably overtake
audio in the jitter buffer. Since the jitter buffer is probably not that
big it's probably not a big deal for most applications.
But I think that not calling MpJbeFixed::update() for DTMFs from
MprDecode::pushPacket()
is a cleaner and easier fix. Would you give it a try and see if that
fixes your
problem as well?
Keith Kyzivat wrote:
> Might you have a patch file you can send up so this can be reviewed
and checked into the repository?
Not quite yet, but I will send it once I did some more testing.
...Carsten
Index: sipXtapi-3.2/sipXmediaLib/src/mp/MprDecode.cpp
===================================================================
--- sipXtapi-3.2/sipXmediaLib/src/mp/MprDecode.cpp (revision 10928)
+++ sipXtapi-3.2/sipXmediaLib/src/mp/MprDecode.cpp (working copy)
@@ -219,11 +219,17 @@
// Initialize JB estimator.
mpJbEstimationState->init(mStreamState.sampleRate);
- // Update jitter state data
- mpJbEstimationState->update(&pRtp->getRtpHeader(),
- mStreamState.rtpStreamPosition,
- mStreamState.playbackStreamPosition,
- &mStreamState.rtpStreamHint);
+ // DTMF telephone-events may have a different rtp->timestamp
+ // and mess up the jitter state calculation, causing audio and
+ // DTMF to stop working.
+ if (!pDecoderInfo->isSignalingCodec())
+ {
+ // Update jitter state data
+ mpJbEstimationState->update(&pRtp->getRtpHeader(),
+ mStreamState.rtpStreamPosition,
+ mStreamState.playbackStreamPosition,
+ &mStreamState.rtpStreamHint);
+ }
mStreamState.rtpStreamPosition += mStreamState.rtpStreamHint;
@@ -234,11 +240,17 @@
// Sample rate mustn't change during the live stream.
assert(pDecoderInfo->getSampleRate() == mStreamState.sampleRate);
- // Update jitter state data
- mpJbEstimationState->update(&pRtp->getRtpHeader(),
- mStreamState.rtpStreamPosition,
- mStreamState.playbackStreamPosition,
- &mStreamState.rtpStreamHint);
+ // DTMF telephone-events may have a different rtp->timestamp
+ // and mess up the jitter state calculation, causing audio and
+ // DTMF to stop working.
+ if (!pDecoderInfo->isSignalingCodec())
+ {
+ // Update jitter state data
+ mpJbEstimationState->update(&pRtp->getRtpHeader(),
+ mStreamState.rtpStreamPosition,
+ mStreamState.playbackStreamPosition,
+ &mStreamState.rtpStreamHint);
+ }
}
RTL_EVENT(str_fg+"_MprDecode_pushPacket_rtp_stream_hint",
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