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List: serusers
Subject: Re: [SR-Users] Rtpengine vs. TURN?
From: Daniel-Constantin Mierla <miconda () gmail ! com>
Date: 2014-07-25 9:13:05
Message-ID: 53D21FA1.5070007 () gmail ! com
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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@gmail.com <mailto:miconda@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
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> list
> sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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Hello,<br>
<br>
<div class="moz-cite-prefix">On 14/07/14 15:49, Peter Villeneuve
wrote:<br>
</div>
<blockquote
cite="mid:CAEiJqQNeT1OEBwmsS00a_oZ97RAEfXhpHQYqJPNNyiJwV72iGQ@mail.gmail.com"
type="cite">
<div dir="ltr">Hi Daniel,
<div><br>
</div>
<div>Thanks for your input. Since I couldn't decide which one to
use, I've been experimenting with using both.</div>
<div>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.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>What's the best way to do this?</div>
</div>
</blockquote>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<blockquote
cite="mid:CAEiJqQNeT1OEBwmsS00a_oZ97RAEfXhpHQYqJPNNyiJwV72iGQ@mail.gmail.com"
type="cite">
<div dir="ltr">
<div><br>
</div>
<div>Cheers,</div>
<div>Peter</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Jul 14, 2014 at 2:18 PM,
Daniel-Constantin Mierla <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:miconda@gmail.com"
target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,
<div>
<div class="h5"><br>
<br>
On 12/07/14 19:55, Peter Villeneuve wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
<br>
On my server, I have the option of using either
Rtpengine for NAT traversal or pure TURN without
rtpengine.<br>
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.<br>
<br>
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?<br>
<br>
</blockquote>
</div>
</div>
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.<br>
<br>
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).<br>
<br>
Cheers,<br>
Daniel<span class="HOEnZb"><font color="#888888"><br>
<br>
-- <br>
Daniel-Constantin Mierla - <a moz-do-not-send="true"
href="http://www.asipto.com" \
target="_blank">http://www.asipto.com</a><br> <a moz-do-not-send="true"
href="http://twitter.com/#%21/miconda" \
target="_blank">http://twitter.com/#!/miconda</a>
- <a moz-do-not-send="true"
href="http://www.linkedin.com/in/miconda"
target="_blank">http://www.linkedin.com/in/miconda</a><br>
<br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list<br>
<a moz-do-not-send="true"
href="mailto:sr-users@lists.sip-router.org"
target="_blank">sr-users@lists.sip-router.org</a><br>
<a moz-do-not-send="true"
href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</font></span></blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" \
href="http://www.asipto.com">http://www.asipto.com</a> <a \
class="moz-txt-link-freetext" \
href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a \
class="moz-txt-link-freetext" \
href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a></pre>
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