[asterisk-users] Fwd: RTP stats explaination

[Arif Hossain]

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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 over this :)

We are getting very poor quality of voice during testing of a new
filtering application of us.

The application receives packets from kernel using netfilter_queue
library. Then insert the packets into a new user managed queue and
does some transformations on it, like concatenation of udp payload.

The network is healthy. Its inside our lab. And it does not drop
packets or anything .

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.