John Melody
2006-Sep-07 02:51 UTC
[asterisk-users] Configuring new IAX2 Jitter Buffer for IVR application.
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.