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 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080409/2b022efd/attachment.html
