<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">I think #webrtc is all the rage for all the good or wrong reasons :-)<div><br></div><div>Is indeed the wrong expectation that a sip server would need to handle this natively but people ask about this and other solutions are there to fill up the gap.<br><div><div><br></div><div>Adrian<br><div><br></div><div><div><div>On 17 Jun 2014, at 13:17, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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<div class="moz-cite-prefix"><tt>Adrian,<br>
<br>
We tried all the time to guide the opensips development (as
project) based on the community needs - basically you add
features on demand/usage - you mentioned you felt like "left
behind feature-wise" - could you mention the features you are
missing (especially that you are a foundation member, and we
should provide guidance for the project). I'm all ears :).<br>
<br>
It is more or less what I'm doing (as user) with the rtpproxy
project - I have the need for some missing features and I'm
asking about the future plan.<br>
<br>
Of course, there must be an understanding that different people
doing different things may have different needs - this is the
beauty of an Open Source project - different people, different
needs, all combined into a unitary effort.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com/">http://www.opensips-solutions.com</a></pre>
On 13.06.2014 20:55, <a class="moz-txt-link-abbreviated" href="mailto:ag@ag-projects.com">ag@ag-projects.com</a> wrote:<br>
</div>
<blockquote cite="mid:AFE6DE57-D8CF-44AA-837B-8BFCEC44EFF4@ag-projects.com" type="cite">
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Guys,
<div><br>
</div>
<div>All these softwares are mature with many years in service
both for the media relays and the SIP part. They deal find with
most of the expected failures, which is what the customers
expect. For the un-expected failures, well the sky if the limit
for optimising with infinite cost/benefit ratio. I personally
did not hear my customers asking for any more resilience or
scalability for the media relay component, so I stopped
optimising long time ago.</div>
<div><br>
</div>
<div>A better question is where would OpenSIPS project go next,
beyond optimisations, as the outside world does not stay still
and the perception of some of my customers is that we are being
left behind feature-wise.</div>
<div><br>
</div>
<div>Adrian</div>
<div><br>
</div>
<div>
<div>
<div>On 13 Jun 2014, at 14:18, Bogdan-Andrei Iancu <<a moz-do-not-send="true" href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>>
wrote:</div>
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<div class="moz-cite-prefix"><tt>Hi Maxim,<br>
<br>
It is good to know about the rtp_cluster, but aside
simplifying things, it does not bring any new
functionality - the LB and failover between RTPproxy
nodes can be done now in OpenSIPS module .<br>
The most challenging thing we are looking at is the
ability to move calls between different instances of
RTPP (for HA purposes)..or some restart persistence
for the sessions - without something like that it's
very hard to deal with SW/HW failures ; there are ways
to go around for scheduled stops/restarts
(maintenance), but non for unexpected failures. <br>
<br>
Thanks and Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com/">http://www.opensips-solutions.com</a></pre>
On 13.06.2014 00:36, Maxim Sobolev wrote:<br>
</div>
<blockquote cite="mid:CAH7qZft=CmyO7nFMsjxCiHQFQBF8e=rEg5SWNRFHKVWEUmhmxw@mail.gmail.com" type="cite">
<div dir="ltr">
<div>Brett, on the HA/carrier-grade side there is
little-advertized middle-layer component called
"rtp_cluster", which in essence is load-balancing,
transparent dispatcher that can be inserted in
between some call-controlling component like
OpenSIPS or Sippy B2BUA and bunch of RTPP instances
running on the same or multiple nodes. From the
point of view of that OpenSIPS it's just another
RTPP instance.<br>
<br>
And it handles all logic necessary to load-balance
incoming requests between online instances plus it
can handle dynamic re-confiduration of the cluster
and track individual nodes going up and down. The
code is pretty usable, we have it deployed for
several customers and it's being actively developed
as well. We have it working reliably controlling up
to 30-40 RTPP instances scattered over at least 5
nodes.<br>
<br>
<a moz-do-not-send="true" href="http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/">http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/</a><br>
<br>
</div>
We have at least one pretty well known service
provider whose name starts with capital V using it in
combination with OpenSIPS to load balance RTP traffic
via bunch of Amazon EC2 instances.<br>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Tue, May 27, 2014 at 6:52
AM, Brett Nemeroff <span dir="ltr"><<a moz-do-not-send="true" href="mailto:brett@nemeroff.com" target="_blank">brett@nemeroff.com</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">Just wanted to add my 0.02 here..
<div><br>
</div>
<div>I totally agree with Bogdan. For the
applications where opensips + a RTP relay make
sense, HA and persistence are much more
important. </div>
<div><br>
</div>
<div>WebRTC and ICE are kinda applications in of
themselves. And although these applications
are going to grow in popularity, the "legacy"
needs for an RTP relay are still massively
prevalent in the space. A general push towards
"Carrier Grade", resiliency and redundancy I
think is much better for the project as a
whole. </div>
<div><br>
</div>
<div>Not only that, consider that applications
requiring ICE or WebRTC will greatly benefit
from HA / persistence, but not so much the
other way around :) </div>
<div><br>
</div>
<div>YMMV</div>
<span class="HOEnZb"><font color="#888888">
<div> <br>
</div>
<div>-Brett</div>
<div><br>
</div>
</font></span></div>
<div class="HOEnZb">
<div class="h5">
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Sun, May 25,
2014 at 6:30 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a moz-do-not-send="true" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px
#ccc solid;padding-left:1ex">
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<div><tt>Hello,<br>
<br>
As always, the truth is in the
middle.<br>
<br>
I agree RTPP is behind on certain
things (and this is why we want to
do them), but on the other hand it
is a good platform with other good
features (missing on the other
relays). RTPP has better ability
in individually controlling the
stream (audio /video), ability to
set timeouts and onhold with no
conflicts, ability to generates
events on timeout, more
flexibility in handling symmetric
/ asymmetric NATs, ability to do
media injection (playback),
ability to do call recording<br>
<br>
What neither mediaproxy, nor
rtpengine have is a mechanism for
implementing RTP failover (for
ongoing calls) or restart
persistence . This is something we
want to look into. I would love to
have ICE and WebRTC on my media
relay, for the HA and persistence
are more important I would say.<br>
<br>
Regards,<br>
</tt>
<div>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com/" target="_blank">http://www.opensips-solutions.com</a></pre>
</div>
<div>
<div> On <a moz-do-not-send="true" href="tel:24.05.2014%2001" value="+12405201401" target="_blank">24.05.2014 01</a>:59,
Muhammad Shahzad Shafi wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div><p>To be honest, i have stopped
using rtpproxy for over 2
years now. It is not evolving
as fast as it should be,
specially in the context of
ICE and WebRTC technologies.</p><p>I would like to suggest that
opensips team should consider
adding support for rtpengine
from SIPWise,</p><p><a moz-do-not-send="true" href="https://github.com/sipwise/rtpengine" target="_blank">https://github.com/sipwise/rtpengine</a></p><p>For now mediaproxy from AG
Projects is the only good
choice for handling media in
opensips with ICE support
(though it still lacks WebRTC
features).</p><p>Thank you.</p>
<div> <br class="webkit-block-placeholder">
</div><p>On 2014-05-23 14:55,
Bogdan-Andrei Iancu wrote:</p>
<blockquote type="cite" style="padding-left:5px;border-left:#1010ff
2px
solid;margin-left:5px;width:100%"><tt>Going
for a public exposure on
this question to Maxim,
maybe we will get an answer
here.<br>
<br>
</tt>
<div><br>
-------- Original Message
--------
<table border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:</th>
<td>RTPproxy project</td>
</tr>
<tr>
<th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:</th>
<td>Mon, 14 Apr 2014
15:03:31 +0300</td>
</tr>
<tr>
<th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:</th>
<td>Bogdan-Andrei
Iancu</td>
</tr>
<tr>
<th align="RIGHT" nowrap="nowrap" valign="BASELINE">To:</th>
<td>Maxim Sobolev</td>
</tr>
<tr>
<th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC:</th>
<td>Razvan Crainea</td>
</tr>
</tbody>
</table>
<br>
<br>
<pre>Hello Maxim,
Long time, no talks, but I hope everything is fine on your side.
I'm reaching you in order to ask about your future plans in regards to
the rtpproxy project? We see no much activity around it and other media
relays are popping around.
RTPP is an essential component for us, we invested a lot of work, we
have many patches (extensions) for it (which we want to push to the
public tree, but there is no answer on this) and we are also looking for
investing a lot into big future plans (as adding more functionalities).
Now, my question is - what is your commitment and disponibility for the
RTPP project ? depending on that we what to re-position ourselves, as we
do not want to waste time and work on things which are out of control.
Best regards,
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com/" target="_blank">http://www.opensips-solutions.com</a>
</pre>
</div>
</blockquote>
<div>
<pre>--
Mit freundlichen Grüßen
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: <a moz-do-not-send="true" href="tel:%2B49%20176%2099%2083%2010%2085" value="+4917699831085" target="_blank">+49 176 99 83 10 85</a>
MSN: <a moz-do-not-send="true" href="mailto:shari_786pk@hotmail.com" target="_blank">shari_786pk@hotmail.com</a>
Email: <a moz-do-not-send="true" href="mailto:shaheryarkh@googlemail.com" target="_blank">shaheryarkh@googlemail.com</a></pre>
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-- <br>
<div dir="ltr">Maksym Sobolyev<br>
Sippy Software, Inc.<br>
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Tel (Canada): +1-778-783-0474<br>
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