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Thanks again, Daniel, that is very helpful.<br>
<br>
BTW, I also found that even without disabling rtpproxy, most data
would be transfered directly between endpoints.<br>
Following is the bandwidth test result of video call between two
Jitsi clients.<br>
<br>
<img alt="rtpproxy enabled"
src="cid:part1.02050405.04010501@gmail.com" height="284"
width="334"><br>
<br>
Figure 1. use_rtpproxy set as "Always with plain SDP"<br>
<br>
<br>
<br>
<img alt="rtpproxty disabled"
src="cid:part2.03070206.03040805@gmail.com" height="281"
width="354"><br>
<br>
Figure 2. use_rtpproxy set as "Never"<br>
<br>
<br>
<div class="moz-cite-prefix">On 2014年10月27日 01:41, Daniel Grotti
wrote:<br>
</div>
<blockquote cite="mid:20141026174152.3C5E6649B3@mail.sipwise.com"
type="cite">
<p dir="ltr">Hi,<br>
Yes disabling rtpproxy rtp will go directly from endpoint to
endpoint, if they can see each others. If you have endpoints
behind nat you need to use rtpproxy.</p>
<p dir="ltr">To disable it you just need to do it via web
interface, on domain/peer level under preferences you can find
the parameter use_rtpproxy. Just set it to 'never'.</p>
<p dir="ltr">But be careful if you have clients behind nat.</p>
<p dir="ltr">Daniel<br>
</p>
<div class="quote">On 26 Oct 2014 16:59, yanzs
<a class="moz-txt-link-rfc2396E" href="mailto:danrenjian@gmail.com"><danrenjian@gmail.com></a> wrote:<br type="attribution">
<blockquote class="quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<meta content="text/html; charset=utf-8"
http-equiv="Content-Type">
thanks, Daniel<br>
<br>
I spend a few days studying concepts like RTP, SIP and VOIP. <br>
<br>
So if I disable 'rtpproxy', RTP stream can be sent directly
between endpoints (or at least without any intermediate server
I owned), it that correct? I am concerned with that because
many concurrent video calls would require a high bandwidth on
the server, and that is a bit expensive.<br>
<br>
<b>And another question</b>, in the "Platform Architecture"
chapter of SPCE handbook, it says " The mediaproxy
configuration file is
/etc/ngcp-config/templates/etc/default/ngcp-mediaproxy-ng-daemon"
which I believe is where I can disable rtpproxy. But I could
not find that "ngcp-mediaproxy-ng-daemon" path with spce 3.4.2
installed by virtual box image, do I need to install spce
using NGCP install CD to get this ngcp-mediaproxy-ng-daemon
service ?<br>
<br>
Thanks in advance.<br>
Russell<br>
<br>
<div class="moz-cite-prefix">On 2014年10月22日 16:10, Daniel
Grotti wrote:<br>
</div>
<blockquote cite="mid:54476665.2090900@sipwise.com"
type="cite">
<meta content="text/html; charset=utf-8"
http-equiv="Content-Type">
<div class="moz-cite-prefix">Hi,<br>
RTP stream (audio/video) just pass through NGCP, which is
just an RTP relay.<br>
You can even bypass NGCP during RTP session, by disable
'rtpproxy' if you want. <br>
It depends what do you need, if you need to perform some
actions on rtp stream (rtptimeout for example), you need
NGCP staying in the middle.<br>
<br>
Daniel<br>
<br>
<br>
<br>
<br>
On 10/22/2014 09:38 AM, yanzs wrote:<br>
</div>
<blockquote cite="mid:54475EE8.3090301@gmail.com"
type="cite">
<meta http-equiv="content-type" content="text/html;
charset=utf-8">
Hi, community<br>
<br>
Our team are building an application for users to make
video call to each other (just for confirming their face
and do some stuff in the system), and we are new to spce
and even voip.<br>
In the sipwise site, it claims that with these recommended
hardware: <br>
<br>
<meta http-equiv="content-type" content="text/html;
charset=utf-8">
<ul style="margin: 0px 0px 10px 0.4em; padding: 0px;
border: 0px; outline: 0px; font-size: 11px;
vertical-align: baseline; list-style: disc; color:
rgb(51, 51, 51); font-family: Arial; font-style: normal;
font-variant: normal; font-weight: normal;
letter-spacing: normal; line-height: 20.7999992370605px;
orphans: auto; text-align: left; text-indent: 0px;
text-transform: none; white-space: normal; widows: auto;
word-spacing: 0px; -webkit-text-stroke-width: 0px;
background: rgb(246, 246, 246);">
<li style="margin: 0px; padding: 0px; border: 0px;
outline: 0px; font-size: 11px; vertical-align:
baseline; list-style: disc inside none; display:
block; background: transparent;">Dual-core x86_64
compatible</li>
<li style="margin: 0px; padding: 0px; border: 0px;
outline: 0px; font-size: 11px; vertical-align:
baseline; list-style: disc inside none; display:
block; background: transparent;">3GHz, 4GB RAM, 128GB
HDD</li>
</ul>
<p>Sipwise can sustain <span style="color: rgb(51, 51,
51); font-family: Arial; font-size: 11px; font-style:
normal; font-variant: normal; font-weight: normal;
letter-spacing: normal; line-height:
20.7999992370605px; orphans: auto; text-align: left;
text-indent: 0px; text-transform: none; white-space:
normal; widows: auto; word-spacing: 0px;
-webkit-text-stroke-width: 0px; display: inline
!important; float: none; background-color: rgb(246,
246, 246);">2.000 Concurrent Calls, </span><span
style="color: rgb(51, 51, 51); font-family: Arial;
font-size: 11px; font-style: normal; font-variant:
normal; font-weight: normal; letter-spacing: normal;
line-height: 20.7999992370605px; orphans: auto;
text-align: left; text-indent: 0px; text-transform:
none; white-space: normal; widows: auto; word-spacing:
0px; -webkit-text-stroke-width: 0px; display: inline
!important; float: none; background-color: rgb(246,
246, 246);">50 Call Attempts per Second.</span><br>
So I am wondering how audio and video stream data are
transfered from one client to another, does clients make
a p2p connection, or all data are passed through the
spce server?<br>
</p>
<p>Thanks, <br>
Russell<br>
<span style="color: rgb(51, 51, 51); font-family: Arial;
font-size: 11px; font-style: normal; font-variant:
normal; font-weight: normal; letter-spacing: normal;
line-height: 20.7999992370605px; orphans: auto;
text-align: left; text-indent: 0px; text-transform:
none; white-space: normal; widows: auto; word-spacing:
0px; -webkit-text-stroke-width: 0px; display: inline
!important; float: none; background-color: rgb(246,
246, 246);"></span></p>
<br>
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