Hi,
Am I correct, that this is a better version of the patch, offered
by Paulo?
On Sat, Nov 15, 2008 at 3:07 AM, Carsten Avenhaus <[EMAIL PROTECTED]> wrote:
> Here is a patch that re-initializes the RTP session and jitter buffer
> after new RTP streams are negotiated, for instance after a supervised
> transfer etc.
>
> ...Carsten
>
>
>
> Index: sipXtapi-3.2/sipXmediaLib/include/mp/MprDejitter.h
> ===================================================================
> --- sipXtapi-3.2/sipXmediaLib/include/mp/MprDejitter.h (revision 10928)
> +++ sipXtapi-3.2/sipXmediaLib/include/mp/MprDejitter.h (working copy)
> @@ -109,6 +109,9 @@
> * amount of time, possible forever.
> */
>
> + /// Reset dejitter to initial state and prepare for new stream.
> + void reset();
> +
> //@}
>
> /* ============================ ACCESSORS =================================
> */
> @@ -148,8 +151,6 @@
> /// we can distinguish out-of-order packets.
> RtpSeq mMaxPulledSeqNo;
>
> - /// Reset dejitter to initial state and prepare for new stream.
> - void reset();
>
> /* //////////////////////////// PRIVATE ///////////////////////////////////
> */
> private:
> 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)
> @@ -555,6 +555,13 @@
> osPrintf("MprDecode::handleSelectCodecs(%d codec%s):\n",
> numCodecs, ((1 == numCodecs) ? "" : "s"));
> #endif
> +
> + // The RTP session and jitter buffer need to be reset
> + // when new RTP streams are negotiated, for instance after a
> + // supervised transfer
> + mIsStreamInitialized = FALSE;
> + mpMyDJ->reset();
> +
> if (OsSysLog::willLog(FAC_MP, PRI_DEBUG))
> {
> for (i=0; i<numCodecs; i++) {
>
> _______________________________________________
> sipxtapi-dev mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
>
--
Regards,
Alexander Chemeris.
SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000
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