<div dir="ltr"><div><div><div>The current target is often places not connected to the internet, but supported by mesh. <br><br></div>It seemed easy to enable AUTH. (just adding #!define WITH_AUTH) to the conf file seemed to do the job. <br><br></div>Kamailio also has a supported web interface system (siremis) that allows for user registration. <br><br></div>Speaking of which, has anyone tried H.323? Thoughts on that?<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Oct 15, 2016 at 6:31 AM, Samuel Greenfeld <span dir="ltr"><<a href="mailto:samuel@greenfeld.org" target="_blank">samuel@greenfeld.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>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.<br><br></div><div>SIP also has some firewall issues, although those can be mitigated with proxies, a STUN server, and the proper configuration.<br></div><div><br></div>IAX2 historically has been meant to be an "Inter-Asterisk" server trunking protocol, not something for individual clients.<br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Oct 14, 2016 at 8:55 PM, Anish Mangal <span dir="ltr"><<a href="mailto:anishmg@umich.edu" target="_blank">anishmg@umich.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div>This looks interesting, but I would prefer to stick with SIP<br><br></div>Reason:<br></div>1. It is extremely scalable from running on tiny openWRT routers to big servers<br></div>2. It is supported by a large number of free/open clients on ALL platforms. There is also a webRTC socket for it.<br></div><br></div>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 <br><div><div><br></div></div></div><div class="gmail_extra"><div><div class="m_-8380591828829383727h5"><br><div class="gmail_quote">On Sat, Oct 15, 2016 at 6:08 AM, Joel Steres <span dir="ltr"><<a href="mailto:joel.steres@ymobility.com" target="_blank">joel.steres@ymobility.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Does webrtc fit your use case? <a href="http://peerjs.com" target="_blank">http://peerjs.com</a> is one.</p><div class="m_-8380591828829383727m_5599765799862117559HOEnZb"><div class="m_-8380591828829383727m_5599765799862117559h5">
<div class="gmail_extra"><br><div class="gmail_quote">On Oct 14, 2016 5:12 PM, "Anish Mangal" <<a href="mailto:anishmg@umich.edu" target="_blank">anishmg@umich.edu</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Following from:<br><a href="http://www.en.voipforo.com/IAX/IAXvsSIP.php" target="_blank">http://www.en.voipforo.com/IAX<wbr>/IAXvsSIP.php</a><br><br><strong>If SIP is using a server</strong> signaling messages always pass through the server but <strong>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. </strong>This increases the bandwidth need by the IAX servers when there are many simultaneous calls.<strong><br>
<br><br></strong></div>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. <br><br></div>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. :)<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Oct 15, 2016 at 5:07 AM, Anish Mangal <span dir="ltr"><<a href="mailto:anishmg@umich.edu" target="_blank">anishmg@umich.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Will look into IAX2. Is it supported by apps on different clients? For SIP, there are usually many client options available on various platforms.<br></div><div class="gmail_extra"><div><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388h5"><br><div class="gmail_quote">On Fri, Oct 14, 2016 at 9:22 PM, Sameer Verma <span dir="ltr"><<a href="mailto:sverma@sfsu.edu" target="_blank">sverma@sfsu.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">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.</p><span class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388m_-647112329684591509HOEnZb"><font color="#888888">
<p dir="ltr">Sameer</p></font></span><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388m_-647112329684591509HOEnZb"><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388m_-647112329684591509h5">
<div class="gmail_extra"><br><div class="gmail_quote">On Oct 14, 2016 4:48 AM, "Anish Mangal" <<a href="mailto:anishmg@umich.edu" target="_blank">anishmg@umich.edu</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>Hi, <br><br></div>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. <br><br></div>I have already included a SIP server in the upcoming deployment of XSCE and mesh in Spiti, north India. <br><br></div>Thoughts?<br><div><div><div><div><br>Best, <br></div><div><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388m_-647112329684591509m_-1099847860933295132m_-3540975107822173674gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Anish<br></div><div><br><br></div></div></div>
</div></div></div></div></div>
</blockquote></div></div>
</div></div></blockquote></div><br><br clear="all"><br></div></div><span class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388HOEnZb"><font color="#888888">-- <br><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388m_-647112329684591509gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Anish<br></div><div><br><br></div></div></div>
</font></span></div>
</blockquote></div><br><br clear="all"><br>-- <br><div class="m_-8380591828829383727m_5599765799862117559m_-6478556602294878936m_-3867373454651979388gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Anish<br></div><div><br><br></div></div></div>
</div>
</blockquote></div></div>
</div></div></blockquote></div><br><br clear="all"><br></div></div><span class="m_-8380591828829383727HOEnZb"><font color="#888888">-- <br><div class="m_-8380591828829383727m_5599765799862117559gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Anish<br></div><div><br><br></div></div></div>
</font></span></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Anish<br></div><div><br><br></div></div></div>
</div>