[SR-Users] Rtpengine vs. TURN?

[Daniel-Constantin Mierla]

Hello,

On 14/07/14 15:49, Peter Villeneuve wrote:
> Hi Daniel,
>
> Thanks for your input. Since I couldn't decide which one to use, I've 
> been experimenting with using both.
> The problem with my mixed approach is that there are too many ICE 
> candidates created (I counted 10 in the last logs I looked at for one 
> call), real relay candidates (turn), and fake host candidates 
> (rtpengine) with different priorities which leads to all kinds of 
> problems.
>
> I think I'll stick to TURN since my clients have support for it. 
> Still, I'd like to keep using the NAT traversal (or more accurately 
> NAT detection) support of Kamailio, but I don't want rtpproxy-ng to 
> add any ICE candidates at all. The reason I need some NAT support in 
> Kamailio is that although most of my clients support ICE/STUN/TURN, 
> others use Jitsi which has no support for these protocols, and I need 
> a way to connect to Jitsi clients that register from behind NAT.
>
> What's the best way to do this?

you can keep rtpproxy in kamailio.cfg. If there is a turn server, the 
SDP should come with a public IP in sdp and then you don't engage the 
rtpproxy -- iirc, the rtpproxy or nathelper module has a test to check 
if the media ip in sdp is a private address. you can use that for 
deciding to do rtp relaying on server or not.

Cheers,
Daniel
>
> Cheers,
> Peter
>
>
> On Mon, Jul 14, 2014 at 2:18 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>
>     On 12/07/14 19:55, Peter Villeneuve wrote:
>
>         Hi,
>
>         On my server, I have the option of using either Rtpengine for
>         NAT traversal or pure TURN without rtpengine.
>         Rtpengine has the obvious plus that it only needs 1 public IP,
>         while TURN (with STUN) will need 2 public IPs, although that's
>         not a problem in my case.
>
>         Having said that, I'd like to take advantage of the huge
>         experience that users of this list have in real world
>         deployments. in your experience, which option is more reliable
>         in a real world deployment?
>
>     TURN is a more standard way, but it requires support in the client
>     implementation and not many of the (rather old) sip hardphones
>     don't support that.
>
>     A RTP relay (like rtpengine, rtpproxy) is server only solution,
>     not requiring anything in the client side. On the other hand is an
>     exposure to less privacy if you don't encrypt the rtp (just
>     because the server controls where to send media).
>
>     Cheers,
>     Daniel
>
>     -- 
>     Daniel-Constantin Mierla - http://www.asipto.com
>     http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
>     http://www.linkedin.com/in/miconda
>
>
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>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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