[Pkg-voip-commits] [janus] 139/282: Clarified that the SIPre demo is not currently as stable as the older Sofia one
Jonas Smedegaard
dr at jones.dk
Wed Dec 20 21:53:35 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a commit to annotated tag debian/0.2.6-1
in repository janus.
commit f40ab2dc671e837bc642e1352cdd89bc11fa7e30
Author: Lorenzo Miniero <lminiero at gmail.com>
Date: Fri Oct 20 15:37:24 2017 +0200
Clarified that the SIPre demo is not currently as stable as the older Sofia one
---
html/demos.html | 15 ++++++++++-----
html/navbar.html | 5 +++--
html/sipretest.html | 11 ++++-------
3 files changed, 17 insertions(+), 14 deletions(-)
diff --git a/html/demos.html b/html/demos.html
index 1af2b59..7e51635 100644
--- a/html/demos.html
+++ b/html/demos.html
@@ -53,10 +53,6 @@
<td>A SIP Gateway demo, allowing you to register at a SIP server and start/receive calls.</td>
</tr>
<tr>
- <td><a href="sipretest.html">SIP Gateway (libre)</a></td>
- <td>Same as the above, but using the libre-based plugin instead of the Sofia-based one.</td>
- </tr>
- <tr>
<td><a href="videoroomtest.html">Video Room</a></td>
<td>A videoconferencing demo, allowing you to join a video room with up to six users.</td>
</tr>
@@ -83,7 +79,16 @@
</table>
<table class="table table-striped">
<tr>
- <td colspan=2><h3>Other demos</h3></td>
+ <td colspan=2><h3>Other legacy demos</h3></td>
+ </tr>
+ <tr>
+ <td><a href="sipretest.html">SIP Gateway (libre)</a></td>
+ <td>Same as the SIP demo above, but using the libre-based plugin instead of the Sofia-based one.</td>
+ </tr>
+ </table>
+ <table class="table table-striped">
+ <tr>
+ <td colspan=2><h3>Advanced demos</h3></td>
</tr>
<tr>
<td><a href="devicetest.html">Device Selection</a></td>
diff --git a/html/navbar.html b/html/navbar.html
index 24da7ff..9ec12e9 100644
--- a/html/navbar.html
+++ b/html/navbar.html
@@ -17,8 +17,7 @@
<li><a href="echotest.html">Echo Test</a></li>
<li><a href="streamingtest.html">Streaming</a></li>
<li><a href="videocalltest.html">Video Call</a></li>
- <li><a href="siptest.html">SIP Gateway (Sofia)</a></li>
- <li><a href="sipretest.html">SIP Gateway (libre)</a></li>
+ <li><a href="siptest.html">SIP Gateway</a></li>
<li><a href="videoroomtest.html">Video Room</a></li>
<li><a href="audiobridgetest.html">Audio Room</a></li>
<li><a href="textroomtest.html">Text Room</a></li>
@@ -26,6 +25,8 @@
<li><a href="recordplaytest.html">Recorder/Playout</a></li>
<li><a href="screensharingtest.html">Screen Sharing</a></li>
<li class="divider"></li>
+ <li><a href="sipretest.html">SIP Gateway (libre)</a></li>
+ <li class="divider"></li>
<li><a href="devicetest.html">Device Selection</a></li>
<li><a href="vp9svctest.html">VP9-SVC Video Room</a></li>
<li class="divider"></li>
diff --git a/html/sipretest.html b/html/sipretest.html
index 6a58543..ed222c3 100644
--- a/html/sipretest.html
+++ b/html/sipretest.html
@@ -59,13 +59,10 @@
This will allow you to call SIP URIs, or receive calls through the SIP Server itself.
During a call, you'll also be able to interact with the PBX via DTMF tones, e.g.,
to drive an Interactive Voice Response (IVR) menu that you're being presented with.</p>
- <div class="alert alert-info"><b>Note well!</b> This demo, as the plugin it makes use of,
- has a few issues, right now. Specifically, it has only been tested with the widespread
- Asterisk PBX (extended with our
- <a href="https://github.com/meetecho/asterisk-opus" target="_blank">Opus and VP8 support</a> patch),
- even though it should work fine with others as well. Besides, video has not been
- tested yet and as such may not work as expected.
- </div>
+ <div class="alert alert-info"><b>Note well!</b> This plugin is currently WIP, and so
+ may not always work as expected. Considering the Sofia-based plugin has been around
+ for much longer, and as of today has been used by more people in production, you
+ may want to stick to that if what you're looking for is stability.</div>
<p>Press the <code>Start</code> button above to launch the demo.</p>
</div>
</div>
--
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