[OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

Bogdan-Andrei Iancu bogdan at voice-system.ro
Sun Feb 15 12:25:47 CET 2009


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 <bogdan at 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 <bogdan at voice-system.ro> wrote:
>>>
>>>       
>>>> Hi Olle,
>>>>
>>>> Johansson Olle E wrote:
>>>>
>>>>         
>>>>> 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo:
>>>>>
>>>>>
>>>>>           
>>>>>> 2009/2/10  <julianokyap at 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 at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>         
>>>       
>>     
>
>   




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