Re: [asterisk-users] Fwd: RTP stats explaination

2012-05-19 Thread Arif Hossain
Hi Dave, On Fri, May 18, 2012 at 11:27 PM, Dave Platt dpl...@radagast.org 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

Re: [asterisk-users] Fwd: RTP stats explaination

2012-05-18 Thread Dave Platt
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 nearly

Re: [asterisk-users] Fwd: RTP stats explaination

2012-05-18 Thread Steve Edwards
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

Re: [asterisk-users] Fwd: RTP stats explaination

2012-05-18 Thread Kevin P. Fleming
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