[pjsip] siprtp problem with asterisk

antoine duclaud antoine.duclaud at gmail.com
Wed Apr 9 03:23:37 EDT 2008


So there is no possibilities to change that?

Do you think there is differences(in the results) between direct path
(caller to callee) and via asterisk (caller<=>asterisk<=>callee) for the
quality of the connection?

2008/4/8, Benny Prijono <bennylp at pjsip.org>:
>
> On Tue, Apr 8, 2008 at 2:02 PM, antoine duclaud
> <antoine.duclaud at gmail.com> wrote:
> > Hello,
> >
> > I actually use siprtp with an asterisk server (2 hosts and one server)
> >
> > when I use siprtp on the two hosts directly :
> >
> > ./siprtp....   -i 172.29.197.48          <<host 1 callee
> >  ./siprtp...  -i 172.29.197.73   sip:172.29.197.48   << host caller
> >
> > It's ok, I have results for RX, TX, RTT
> >
> > when I use siprtp via asterisk (172.29.197.104)
> >
> > ./siprtp.... -i 172.29.197.48  << host callee
> > ./siprtp... -i 172.29.197.73   sip:1112 at 172.29.197.48<sip%3A1112 at 172.29.197.48>  <<
> host caller
> >
> >
> > I have results for RX but not for TX and RTT (last update never)
> >
>
>
> Then it looks like Asterisk doesn't support RTCP, since the TX
> statistic is calculated from the RTCP packets received from remote
> endpoint.
>
> Cheers
>   Benny
>
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