The current target is often places not connected to the internet, but
supported by mesh.

It seemed easy to enable AUTH. (just adding #!define WITH_AUTH) to the conf
file seemed to do the job.

Kamailio also has a supported web interface system (siremis) that allows
for user registration.

Speaking of which, has anyone tried H.323? Thoughts on that?

On Sat, Oct 15, 2016 at 6:31 AM, Samuel Greenfeld <sam...@greenfeld.org>
wrote:

> The downside to SIP is that if you leave a SIP server open on the Internet
> which allows calls without authenticated registration (even by accident)
> you will find dozens of systems trying to proxy long-distance &
> international telephone calls through it.  If they can get through,
> thousands of dollars of those calls may be billed to you.
>
> SIP also has some firewall issues, although those can be mitigated with
> proxies, a STUN server, and the proper configuration.
>
> IAX2 historically has been meant to be an "Inter-Asterisk" server trunking
> protocol, not something for individual clients.
>
> On Fri, Oct 14, 2016 at 8:55 PM, Anish Mangal <anis...@umich.edu> wrote:
>
>> This looks interesting, but I would prefer to stick with SIP
>>
>> Reason:
>> 1. It is extremely scalable from running on tiny openWRT routers to big
>> servers
>> 2. It is supported by a large number of free/open clients on ALL
>> platforms. There is also a webRTC socket for it.
>>
>> The only current drawback of Kamailio right now, is there dont seem to be
>> ootb rpms available for arm. If there is a SIP server out there with ARM
>> packages, would love to test it, or otherwise - compile kamailio for ARM
>>
>>
>> On Sat, Oct 15, 2016 at 6:08 AM, Joel Steres <joel.ste...@ymobility.com>
>> wrote:
>>
>>> Does webrtc fit your use case? http://peerjs.com is one.
>>>
>>> On Oct 14, 2016 5:12 PM, "Anish Mangal" <anis...@umich.edu> wrote:
>>>
>>>> Following from:
>>>> http://www.en.voipforo.com/IAX/IAXvsSIP.php
>>>>
>>>> *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. :)
>>>>
>>>> On Sat, Oct 15, 2016 at 5:07 AM, Anish Mangal <anis...@umich.edu>
>>>> wrote:
>>>>
>>>>> Will look into IAX2. Is it supported by apps on different clients? For
>>>>> SIP, there are usually many client options available on various platforms.
>>>>>
>>>>> On Fri, Oct 14, 2016 at 9:22 PM, Sameer Verma <sve...@sfsu.edu> wrote:
>>>>>
>>>>>> What server/service are you using? Look at IAX2 as well. Usually IAX2
>>>>>> does better on networks because unlike SIP, session initiation and voice
>>>>>> call happen on the same port.
>>>>>>
>>>>>> Sameer
>>>>>>
>>>>>> On Oct 14, 2016 4:48 AM, "Anish Mangal" <anis...@umich.edu> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> A SIP server on the XSCE will allow for VOIP services (audio, video,
>>>>>>> text). There are numerous SIP clients on various platforms so it seems a
>>>>>>> good protocol and standard to build upon.
>>>>>>>
>>>>>>> I have already included a SIP server in the upcoming deployment of
>>>>>>> XSCE and mesh in Spiti, north India.
>>>>>>>
>>>>>>> Thoughts?
>>>>>>>
>>>>>>> Best,
>>>>>>> Anish
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anish
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anish
>>>>
>>>>
>>>>
>>
>>
>> --
>> Anish
>>
>>
>>
>


-- 
Anish
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