[asterisk-dev] Asterisk RTP Jitter Buffer and DAHDI Channels

[Kevin P. Fleming]

On 04/25/2012 09:27 AM, Fernando Berretta wrote:
> Hi Everybody,
>
> We recently had some problems with some SIP/PSTN gateways have not
> properly implemented RTP Jitter buffer, and as work around, we have
> forced a Jitter buffer in Asterisk and all worked ok.. but with extra
> delay of course. This situation have maked us think about all possible
> problems related with common RTP Jitter Buffer and how Asterisk handle
> Jitter Buffer when it is working as a SIP/PSTN gateway through DAHDI
> Channels. Could some one there please let me know if asterisk / dahdi
> ,by default, do something about Jitter Buffer with calls which comes
> from VoIP Network and goes to PSTN through Asterisk acting as a gateway

DAHDI itself provides some very basic de-jittering, but it does not do 
packet loss concealment or reordering. DAHDI channels have an 'input 
buffer' that can hold a configurable number of frames of audio, and the 
buffer policy can be configured so that the buffer must be full (or 
half-full) before any audio will be played out to the channel. In this 
configuration, small variations in the packet delivery times over the IP 
network will be absorbed by the DAHDI channel, but as you say, this 
comes at the cost of introducing some delay in the audio path (as all 
jitter buffering does).

Note that the default configuration for DAHDI channels won't help much, 
as the normal mode for the input buffer is 'immediate' mode (audio 
playout begins as soon as the first frame is stored in the buffer). If 
the next frame arrives before the first one is done playing out, then 
the buffer will accept it and it won't have to be dropped, but if the 
next frame is late, then there would be a gap in the playout. The 'half' 
and 'full' buffer policies in DAHDI allow you to decide how you want to 
address this problem.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org