[Kamailio-Users] stun/outbound draft...

[Klaus Darilion]

Aymeric Moizard schrieb:
> 
> On Sun, 4 Jan 2009, Juha Heinanen wrote:
> 
>> Aymeric Moizard writes:
>>
>>> * TURN is used ONLY when 2 peers cannot connect together, this means
>>>    that it's much better than always offering RTP relay which is
>>>    the solution today.
>> kamailio tools allow you to choose when to use rtp relay and when not.
>> it is not used always in my configuration.
> 
> My opinion is that no decision can be taken on kamailio. Or any proxy... 
> It's technically not possible. Only the client and ICE could decide wether 
> to use relay or not.
> 
> All the tools/way/tricks tha I'm aware of, TRY to guess by looking at
> IPs for example, or comparing source port and contact or via port: all
> those tricks are just bad guess for many reasons.
> 
> If you have a 100% working trick, I'll be interested to learn it! Very
> interested!

Another working trick is to use the media relay in 100% of the calls. 
Yes, it is not optimal but it works - and support hours do costs more 
money than bandwidth.

regards
klaus

> 
>>> * ICE allow to certify that you are sending the RTP data to your
>>>    correspondant (ther is an ICE password in SDP)
>> there are also other solutions for this, like quite commonly implemented
>> zrtp, which has the additional advantage that rtp is also encrypted.
> 
> ICE helps to find direct connection between peers securely: still it
> doesn't prevent from receiving spoofed data on the discovered connection.
> zrtp is there to fill the gap: not a conflict at all.
> 
>> negative aspect of ice is that the spec is VERY complicated and hard to
>> understand.  even the tutorial by rosenberg was too much to me.
> 
> Well, I'm not saying it's not complicated... I fully agree there. But
> still this is the best (& only?) way to optimize the SIP udp media path.
> 
> Until ICE come, there will never be any success for the internet part of
> SIP: the only reliable (optimized) calls today are the SIP to PSTN calls
> and that is very very sad and unusefull.
> 
> Of course, non optimized free SIP (telephony..) services can be provided, 
> but this comes at a high cost: you have to provide the relay services, 
> modify contact headers, modify SDP contents, break SIP message integrity, 
> provide low quality voice & video (relay...). Last point but not the 
> least, is there any existing way to make sure you can provide inter-domain
> SIP calls? (2 RTP relays are inserted in the path or a proxy fix a 
> contact header to the IP/port source of the previous proxy...)
> 
> I want ICE to be implemented, no matter the effort. Else, I think there is
> no value in SIP or VoIP except for commercial usage.
> 
> Please help the world and vote for ICE whatever the effort. If not, join
> the IMS effort to make millions of $ for the next decades. That's a 
> choice! (no flame to anyone here, don't misunderstood me... I just 
> feel sad)
> 
> MAIL was a success & free probably because correct deployment is easy.
> With ICE, SIP could become the same.
> 
> I'm done!
> Aymeric MOIZARD / ANTISIP
> amsip - http://www.antisip.com
> osip2 - http://www.osip.org
> eXosip2 - http://savannah.nongnu.org/projects/exosip/
> 
> 
> 
>> -- juha
>>
> 
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