[asterisk-users] Configuring new IAX2 Jitter Buffer for IVR application.

[John Melody]

Hi,

I have a Asterisk configuration as follows

			SIP(LAN)	          IAX2(WAN)
	PSTN ----> GW ------------>  *-client ------------------> *-Server

The *-Server serves recorded prompts as part of an IVR menu to the *-Client

I am using the new JitterBuffer in the *-Client to de-jitter the audio
coming from the server.

The rtt on the WAN is typically 18 - 24ms between the client and server but
occasionally this jumps to 200ms for a short period giving distortion in the
received audio.
The Jitterbuffer debug output shows packets arriving at or around these
times as "L" (for Lost) followed by "l" (for late).

Is it possible to configure the new jitterbuffer as a playback buffer that
introduces a static 500mS delay for example so that the late packets are not
discarded.  The 1/2 second delay introduced by the jitterbuffer is not
really an issue because it is an IVR application. I notice that in the
original JitterBuffer design there was mention of two modes for setting up
the jitterbuffer a JITTERBUFFER_MODE_RECORD  as well as a
JITTERBUFFER_MODE_REALTIME. Is this possible and if so how do you set it up.

Perhaps there is another way to achieve this. Any suggestions would be
appreciated.

regards,
John.