[asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Olli Heiskanen]

Thanks Paul, I appreciate your thoughts.

I understand your way, it's logical in your environment. I prefer to use
LTS versions of Asterisk so I'm guessing what I want to do is not quite
possible with Asterisk 11.

I'd prefer my setup to work like this in different cases.

webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- webrtc (rtp/savpf)
sip (rtp/avp) -- kamailio -- rtpengine (rtp/savpf) -- asterisk -- kamailio
-- rtpengine (rtp/avp) -- sip (rtp/avp)
webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- rtpengine
(rtp/avp) -- sip (rtp/avp)

... essentially, using RTP/AVP only when the client does not speak securely.

It appears I'll have to try out the RTP/AVP way until there is an Asterisk
that can accomplish this without having to use peer-specific settings.
Down-side to this is that rtpengine needs resources from the server for
webrtc clients even though both ends speak the same profile.

It's not so complicated now that I know more on what Asterisk supports and
how it handles the sdp, I just needed to learn by doing, testing and
asking. I must be a bit ahead of my time for going for a RTP/SAVPF within
my architecture, but using RTP/AVP is not such a bad option as srtp is on
its way anyway in future Asterisk versions and the rtp flowing between
Kamailio and users' networks are far more important than internal rtp
traffic.

cheers,
Olli





2014-08-15 18:48 GMT+03:00 Paul Belanger <paul.belanger at polybeacon.com>:

> On Fri, Aug 15, 2014 at 10:41 AM, Olli Heiskanen
> <ohjelmistoarkkitehti at gmail.com> wrote:
> > Hello,
> >
> > After having thought this through a bit I have some thoughts I'd like to
> > share.
> >
> > In this case where the rtp profile is RTP/AVP Asterisk accepts and
> handles
> > the call normally. If a webrtc client calls a sip client, or even another
> > webrtc client, rtpengine is needed to step in (in my setup most of the
> > clients would indeed be webrtc, but some of them might be sip). I think
> it
> > would be better to use RTP/SAVPF throughout the process if both clients
> are
> > webrtc (or otherwise speak RTP/SAVPF), but currently there is no way to
> > accomplish this?
> >
> > Is it possible to configure Asterisk to only accept the RTP/SAVPF
> profile,
> > and send 488 to all others? If it's not possible to force Asterisk to
> ignore
> > rtp profiles (thus allowing the sdp be handled by rtpengine entirely),
> I'd
> > prefer to use RTP/SAVPF or RTP/SAVP in the communication between Kamailio
> > and Asterisk servers and use rtpengine to bridge to RTP/AVP and RTP/AVPF
> > only if the client cannot speak securely.
> >
> > I'd very much like to hear opinions and thoughts on these.
> >
> Again, I'll only share my experiences, but we do the complete
> opposite.  Traffic between kamailio and asterisk is only RTP/AVP since
> the version of asterisk we are using does not support RTP/SAVPF (1.8).
> However, if you want RTP/SAVPF then honestly, you should just
> completely remove rtpengine from the picture since newer version of
> asterisk support both RTP/AVP and RTP/SAVPF (asterisk 12+).
>
> What I think you should do is go back to the basics, and document
> everything you want to do.  Right now you have too many pieces in the
> puzzle and making the setup complicated.  Like I said before, this is
> a complex setup and you need to start some place.  Here is a diagram
> of what we do.
>
> webrtc (RTP/SAVPF) -> kamailio -> rtpengine  -> asterisk (RTP/AVP)
>
> This way, only RTP/AVP is in the core of our network. Rtpengine is on
> the edge (where it belongs), proxing rtp traffic.  And, for us, we
> keep RTP/SAVPF outside of asterisk since support for it has been
> recently added. I also believe there are some open issue with dtls +
> srtp too.
>
> --
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> Github: https://github.com/pabelanger | Twitter:
> https://twitter.com/pabelanger
>
> --
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