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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