Scott Lawrence pisze: > On Thu, 2010-01-28 at 15:10 -0500, Tony Graziano wrote: >> Your performance will be much better on internal call using it with peer to >> peer media. The call quality would be better too. >> >> IF it were me, I'd install a system at each site, so when calls occur >> between site you can enforce using sipxbridge to anchor the media. >> >> Anchoring the media for a call between 2 phones in a single site via the >> server is not necessary and punitive in performance and quality, in general. > > Tony is on the right track if you want to fight with it, but > > Given your network, I'd just say that sipXecs isn't a good candidate. > Indeed, I'd say that SIP, and maybe even VoIP itself isn't a good idea. > > Hi. I'd like to report that i have used ser/kamailio with rtp proxy as outbound proxy for phones. ser is changing Contact header in REGISTERs and passes them to sipx. on INVITE ser is tweaking sdp to engage rtpproxy and pass it to sipx. Everything is sitting on one machine and works fine for now.
I have issue with sipxbridge which seems to reuse rtp sockets from old calls causing randomly one way audio. So i replaced it with some b2bua i cooked using sems. But i think i'd better make issue on jira about it as i have pcap dump with evidence to upload. Grzegorz Stanislawski _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
