For what it's worth, I also got asterisk and freepbx to work (atleast on
centos). It is definitely has more features than kamailio.. and allows for
stuff like voicemail, which can be especially useful in environments with
flakey connectivity. :-)

It might take a bit of time to package as I will be travelling soon, but it
is promising that it is working. :)

On Sun, Oct 16, 2016 at 10:47 AM, Alex Kleider <> wrote:

> On 2016-10-15 13:02, Sameer Verma wrote:
>> On Fri, Oct 14, 2016 at 5:12 PM, Anish Mangal <> wrote:
>>> Following from:
>>> If SIP is using a server signaling messages always pass through the
>>> server
>>> but audio messages (RTP flow) can travel end to end without passing
>>> through
>>> the server. In IAX, signaling and data must pass always through IAX
>>> server.
>>> This increases the bandwidth need by the IAX servers when there are many
>>> simultaneous calls.
>>> This is a big drawback of IAX it seems, especially in a mesh setup,
>>> where in
>>> many cases, the available bandwidth between clients may be higher via
>>> direct
>>> node routes compared with the bandwidth via the server route. It seems
>>> SIP
>>> will utilize the network more efficiently in a mesh topology.
>>> Yesterday we were testing this on the server, and two nodes with three
>>> client. The data was being sent directly client -- node -- node --
>>> client,
>>> and virtually no bandwidth was being used on the server. :)
>> Not always the case. One of my students worked on her thesis where she
>> set up a bunch of XO-1 laptops over a 802.11s (draft) mesh and tested
>> simultaneous calls to look for saturation, etc.
>> 6BI1-jla/tilila-moujahid-thesis.pdf?dl=0
>> I'll see if I can find the actual thesis.
>> Sameer
> Tilila El Moujahid presented her set up at one of the annual olpc-SF
> summits held a few years ago.
> The next summit is coming up soon.

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