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

Olli Heiskanen ohjelmistoarkkitehti at gmail.com
Fri Aug 15 11:17:51 CDT 2014

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


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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