Hi all, I eventually played around with the Audiocodes box and enabled some settings so it worked with Comedia support.
Thanks, Julian On 2/10/09, Bogdan-Andrei Iancu <[email protected]> wrote: > HI Julian, > > If it has, you can actually force it by adding "direction=active" into > SDP as indication. See "fix_nated_sdp("1") : > http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439 > > Regards, > Bogdan > > Julian Yap wrote: >> Thanks all. I'll check to see if the AudioCodes gateway does have >> comedia support. >> >> That clarifies some half baked NAT/RTP knowledge in my head. >> >> - Julian >> >> >> On 2/10/09, Bogdan-Andrei Iancu <[email protected]> wrote: >> >>> Hi Olle, >>> >>> Johansson Olle E wrote: >>> >>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: >>>> >>>> >>>>> 2009/2/10 <[email protected]>: >>>>> >>>>>>> You don't know if RtpProxy should be running, does it mean you are >>>>>>> trying to use it or not? I don't want to spend time inspecting what >>>>>>> you want to do by reading your config, sorry. >>>>>>> >>>>>> Yeah, I'm trying not to run RTPProxy. After more testing, I'm >>>>>> thinking I may >>>>>> need to. >>>>>> >>>>> You cannot decide if you need RtpProxy or not based on testing, it's >>>>> pure theory: >>>>> >>>>> A RTP proxy is NOT needed when (assuming the proxy has in the public >>>>> internet): >>>>> >>>>> - Both caller and callee have public IP or use STUN. >>>>> - Both caller and callee are in the *SAME* private LAN. >>>>> - The caller is in a private LAN and the callee has public IP and >>>>> supports Comedia mode (typical in some media servers and gateways). >>>>> - The callee is in a private LAN and the caller has public IP and >>>>> supports Comedia mode. >>>>> >>>>> >>>>> A RTP proxy is needed when: >>>>> >>>>> - Caller is in private LAN (with no STUN) and callee in public >>>>> internet (and not supporting Comedia). >>>>> - Caller and callee are in different private LAN's with no STUN. >>>>> >>>> I would like to add that it's the device that can't receive audio that >>>> needs the RTP proxy to get incoming audio. >>>> >>>> If both devices are on private IP's, there's going to be two >>>> RTP proxys involved if they're on different SIP networks. >>>> >>>> Each SIP service needs an RTP proxy for supporting their >>>> local users. >>>> >>>> To simplify: >>>> >>>> - If my user is on a private IP and sends an INVITE, add RTP proxy >>>> handling to the INVITE >>>> >>>> - If my user receives a call and sends a 200 OK, add RTP proxy >>>> handling to the 200 OK >>>> >>>> >>> This logic is simple but not efficient....Theoretically, if a call has >>> already a leg in public net, there is not need for a media relay for >>> traversing the nat. >>> >>> The only requirement is that all the devices to support symmetric media >>> (comedia support). >>> >>> So, after the caller proxy forced RTPproxy, the callee should not do the >>> same because the SDP already have a public IP, the nat traversal works >>> even if the callee is behind a nat. >>> >>> Regards, >>> Bogdan >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
