Hi All,
Thanks nanang for the timely reply.I had analysed the call flow with
wireshark and found that the pjsip endpoints were not receiving Rtcp RR
packets that's why all TX and Rtt were shown as zero.Finally also found
that this behaviour was due to asterisk not sending Rtcp RR packets to
pjsip endpoints whenever I use ulaw codec.
In the asterisk--> by using rtcp debug command I found asterisk receives
RTCP SR packets from both the pjsip endpoints and never sends back RTCP
RR packet.But If force asterisk to use codec other ulaw then I see RTCP
RR packets being sent from asterisk to pjsip endpoints.
This looks like some asterisk bug, I have also posted in asterisk
forum,anyways If anyone had experienced this bug before and have a
solution please help me in this.
with regards
raja
>Message: 4
> Date: Wed, 25 Jun 2008 00:34:58 +0700
>From: "Nanang Izzuddin" <nanang@xxxxxxxxx>
>Subject: Re: Rtcp clarification
>To: "pjsip list" <pjsip at lists.pjsip.org>
>Message-ID:
<c7f43120806241034h6137174fge24e62d38f81c946 at mail.gmail.com>
>Content-Type: text/plain; charset="iso-8859-1"
>Hi Raja,
>AFAIK, RTT calculation is based on RTCP packets, nothing to do with the
>codecs. So first, please make sure RTCP traffic is received correctly,
>e.g:
>using wireshark/net sniffing tool.
>Regards,
> nanang
2008/6/24 raja <raja at solnettechnologies.com>:
> Hi all,
>
> I am running pjsip(0.8.0) calls between US and india through
asterisk.I
> am also consecutively monitoring rtcp informations in these
> pjsi[ endpoints.
>
> In this I am always getting all the orig and term RTT as 0.000ms and
> also all orig jitter as 0.000ms and only get term max jitter as
> 1015.000ms.
>
> I am using PCMU codec for the calls,I am not sure why the orig/term
RTT
> are always shown as 0.000ms. please help me in explaining why RTT
values
> are zero.And also how the Orig jitter values are also 0.000ms.
>
> Also note that other codecs like sppex,ilbc are giving non-zero
values.
>
> with regards
> raja
>
>
