Hi Julian, That is cool - in this way you save a lot of bandwidth and processing power with media relaying.
Regards, Bogdan Julian Yap wrote: > 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 <bog...@voice-system.ro> 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 <bog...@voice-system.ro> wrote: >>> >>> >>>> Hi Olle, >>>> >>>> Johansson Olle E wrote: >>>> >>>> >>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: >>>>> >>>>> >>>>> >>>>>> 2009/2/10 <julianok...@gmail.com>: >>>>>> >>>>>> >>>>>>>> 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 >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >> > > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users