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<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">
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charset=windows-1252">
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=windows-1252">
<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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