Re: [Ekiga-list] SIP and RTP

[Fedor vonBock <f vonbock gmail com>]

thanks,

sorry, about "implement on an Asterisk server" I just mean that I'll be using RTP on Asterisk.


I've known the SRTP plug-in on Asterisk, but I also heard something that about SRTP being implemented on ekiga (at least here http://mail.gnome.org/archives/ekiga-devel-list/2006-October/msg00012.html and this was about 4 years ago),     maybe i ought to check the development list

Thanks,

On Wed, Nov 24, 2010 at 4:23 PM, Jānis Rukšāns <thedogfarted gmail com> wrote:

On Wed, Nov 24, 2010 at 8:18 PM, Fedor vonBock <f vonbock gmail com> wrote:
> I'm confused a bit with how  ekiga can be integrated with any SIP server.  I
> just want to know how the Real TIme Protocol is involoved here (since that
> is what I plan to implement on an Asterisk server)

What do you mean with "plan to implement on an Asterisk server"?
Asterisk already "kinda" supports SIP - all you have to do is to
configure the clients and the dial plan.

> I understand that SIP is mainly a negoatiation protocol while RTP is the
> protocol that carries the audio (or vidieo) and I plan to use Secure RTP
> later on.

Well, technically SIP by itself is a signalisation protocol - it
carries around info about call setup, progress and stuff like that,
plus info about the media (audio and video). The SIP standard requires
that every SIP user agent [1] supports SDP (Session Description
Protocol) as the media description language, and an offer/answer model
for negotiating the actual media that will be used during the call
(defined in RFC 2543 [2]). The latter in turn requires that RTP must
be supported as a way to carry the media over the net.

So this is how we get to "Ekiga can be integrated with any SIP server"
(in theory [3]). For the media endpoints, RTP is required due to the
above, and anything in the middle cares only about the SIP stuff.
Asterisk is like a "phone on steroids" and therefore qualifies as an
"endpoint" here, and thus is a subject to the RTP requirement.

As for the SRTP, as far as I know, it is not supported by Ekiga; and
you need a patched Asterisk - it doesn't support SRTP out of the box.


If you plan to set up an Asterisk, I suggest you to read a bit on SIP
and associated protocols to get a glimpse on how they work together.
Even if not strictly required, that knowledge might save you some
headache later.

Cheers

--
Ian

[1] A SIP user agent is a piece of software/hardware that actually
processes the SIP messages, rather than just passing them around. As
SIP is peer-to-peer protocol, the term "server" is ambiguous here, as
most user agents are both clients (they send requests and process
responses) and servers (they process requests and respond to them).
This way, Ekiga is a "server", too. Compare this to HTTP, where a web
browser only sends requests, and a web server only responds to
requests.

[2] There exists more than one negotiation protocol, but the
particular one from RFC 2543 is required to be supported by all SIP
user agents.

[3] Every non-trivial software has bugs, and Ekiga isn't an exception.
Neither is Asterisk. In addition to that, developers have a tendency
to omit some obscure corner cases (it works for the most folks, but
you never know when yours will be the corner case not implemented),
and to interpret the same thing differently (leading to compatibility
issues).
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