[Asterisk-Dev] Re: [Asterisk-Users] does Jitter calculation in chan_iax2.c work???

Vij v.vijayakumar at gmail.com
Fri May 27 05:36:02 MST 2005


SteveK and Andrew,
Thanks a lot for the suggestion. It helped. We didnt know that jitterbuffer 
wont be enabled with sip endpoints. "forcejitterbuffer=true" solved the 
problem.

Thanks again,
Vijay & Ashish

On 5/27/05, Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com> wrote:
> 
> On May 27, 2005 01:47 am, Vij wrote:
> > The above command always shows zero value for jitter. (Actually, only 
> rtt
> > and kpkts are non-zero). The behaviour is the same even for
> > cross-continental calls.
> 
> Post your iax.conf without passwords.
> 
> Also, are there any native bridges going on on either side? There will be 
> no
> jitter buffer used if so. Also, there will be no jitter buffer enabled if
> the endpoints just go to another VOIP technology (e.g. to another IAX 
> phone
> or to a SIP phone).
> 
> You can force it with "forcejitterbuffer=yes" in iax.conf.
> 
> > Is this a bug in the implementation or a configuration problem?.
> 
> Honestly, you haven't even begun to give us enough information to 
> determine
> that.
> 
> -A.
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