Hi Dave,
On Fri, May 18, 2012 at 11:27 PM, Dave Platt wrote:
>
>> In our app we do not forward packet immediately. After enough packet
>> received to increase rtp packetization time (ptime) the we forward the
>> message over raw socket and set dscp to be 10 so that this time
>> packets can escape
On 05/18/2012 12:51 PM, Steve Edwards wrote:
On Fri, 18 May 2012, Dave Platt wrote:
A maximum jitter of 230 milliseconds looks pretty horrendous to me.
This is going to cause really serious audio stuttering on the
receiving side, and/or will force the use of such a long "jitter
buffer" by the r
On Fri, 18 May 2012, Dave Platt wrote:
A maximum jitter of 230 milliseconds looks pretty horrendous to me. This
is going to cause really serious audio stuttering on the receiving side,
and/or will force the use of such a long "jitter buffer" by the receiver
that the audio will suffer from an i
> In our app we do not forward packet immediately. After enough packet
> received to increase rtp packetization time (ptime) the we forward the
> message over raw socket and set dscp to be 10 so that this time
> packets can escape iptable rules.
>
>>From client side the RTP stream analysis shows
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
This question is not related to asterisk, but related to voip quality
in general. But i thought there are lot of experienced guys out here
who can help me with this. And our telephony platform is also asterisk
:). May be i can extract some bias
