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+<html><head><title>ChangeLog for asterisk-22.6.0-rc2</title></head><body>
+<h2>Change Log for Release asterisk-22.6.0-rc2</h2>
+<h3>Links:</h3>
+<ul>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc2.html">Full ChangeLog</a> </li>
+<li><a href="https://github.com/asterisk/asterisk/compare/22.6.0-rc1...22.6.0-rc2">GitHub Diff</a> </li>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-rc2.tar.gz">Tarball</a> </li>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
+</ul>
+<h3>Summary:</h3>
+<ul>
+<li>Commits: 3</li>
+<li>Commit Authors: 1</li>
+<li>Issues Resolved: 3</li>
+<li>Security Advisories Resolved: 0</li>
+</ul>
+<h3>User Notes:</h3>
+<h3>Upgrade Notes:</h3>
+<h3>Developer Notes:</h3>
+<h3>Commit Authors:</h3>
+<ul>
+<li>George Joseph: (3)</li>
+</ul>
+<h2>Issue and Commit Detail:</h2>
+<h3>Closed Issues:</h3>
+<ul>
+<li>1457: [bug]: segmentation fault because of a wrong ari config</li>
+<li>1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.</li>
+<li>1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes</li>
+</ul>
+<h3>Commits By Author:</h3>
+<ul>
+<li>
+<h4>George Joseph (3):</h4>
+</li>
+<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
+<li>chan_websocket: Fix codec validation and add passthrough option.</li>
+<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
+</ul>
+<h3>Commit List:</h3>
+<ul>
+<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
+<li>chan_websocket: Fix codec validation and add passthrough option.</li>
+<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
+</ul>
+<h3>Commit Details:</h3>
+<h4>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</h4>
+<p>Author: George Joseph
+ Date: 2025-09-23</p>
+<p>In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
+ needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
+ AST_RTP_INSTANCE_RTCP_MUX is set.</p>
+<p>Resolves: #1474</p>
+<h4>chan_websocket: Fix codec validation and add passthrough option.</h4>
+<p>Author: George Joseph
+ Date: 2025-09-17</p>
+<ul>
+<li>Fixed an issue in webchan_write() where we weren't detecting equivalent
+ codecs properly.</li>
+<li>Added the "p" dialstring option that puts the channel driver in
+ "passthrough" mode where it will not attempt to re-frame or re-time
+ media coming in over the websocket from the remote app. This can be used
+ for any codec but MUST be used for codecs that use packet headers or whose
+ data stream can't be broken up on arbitrary byte boundaries. In this case,
+ the remote app is fully responsible for correctly framing and timing media
+ sent to Asterisk and the MEDIA text commands that could be sent over the
+ websocket are disabled. Currently, passthrough mode is automatically set
+ for the opus, speex and g729 codecs.</li>
+<li>Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
+ ensure proper translation paths are set up when switching between native
+ frames and slin silence frames. This fixes an issue with codec errors
+ when transcode_via_sln=yes.</li>
+</ul>
+<p>Resolves: #1462</p>
+<h4>res_ari: Ensure outbound websocket config has a websocket_client_id.</h4>
+<p>Author: George Joseph
+ Date: 2025-09-12</p>
+<p>Added a check to outbound_websocket_apply() that makes sure an outbound
+ websocket config object in ari.conf has a websocket_client_id parameter.</p>
+<p>Resolves: #1457</p>
+</body></html>
--- /dev/null
+
+## Change Log for Release asterisk-22.6.0-rc2
+
+### Links:
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc2.html)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0-rc1...22.6.0-rc2)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-rc2.tar.gz)
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
+
+### Summary:
+
+- Commits: 3
+- Commit Authors: 1
+- Issues Resolved: 3
+- Security Advisories Resolved: 0
+
+### User Notes:
+
+
+### Upgrade Notes:
+
+
+### Developer Notes:
+
+
+### Commit Authors:
+
+- George Joseph: (3)
+
+## Issue and Commit Detail:
+
+### Closed Issues:
+
+ - 1457: [bug]: segmentation fault because of a wrong ari config
+ - 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
+ - 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes
+
+### Commits By Author:
+
+- #### George Joseph (3):
+ - res_ari: Ensure outbound websocket config has a websocket_client_id.
+ - chan_websocket: Fix codec validation and add passthrough option.
+ - res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
+
+
+### Commit List:
+
+- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
+- chan_websocket: Fix codec validation and add passthrough option.
+- res_ari: Ensure outbound websocket config has a websocket_client_id.
+
+### Commit Details:
+
+#### res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
+ Author: George Joseph
+ Date: 2025-09-23
+
+ In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
+ needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
+ AST_RTP_INSTANCE_RTCP_MUX is set.
+
+ Resolves: #1474
+
+#### chan_websocket: Fix codec validation and add passthrough option.
+ Author: George Joseph
+ Date: 2025-09-17
+
+ * Fixed an issue in webchan_write() where we weren't detecting equivalent
+ codecs properly.
+ * Added the "p" dialstring option that puts the channel driver in
+ "passthrough" mode where it will not attempt to re-frame or re-time
+ media coming in over the websocket from the remote app. This can be used
+ for any codec but MUST be used for codecs that use packet headers or whose
+ data stream can't be broken up on arbitrary byte boundaries. In this case,
+ the remote app is fully responsible for correctly framing and timing media
+ sent to Asterisk and the MEDIA text commands that could be sent over the
+ websocket are disabled. Currently, passthrough mode is automatically set
+ for the opus, speex and g729 codecs.
+ * Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
+ ensure proper translation paths are set up when switching between native
+ frames and slin silence frames. This fixes an issue with codec errors
+ when transcode_via_sln=yes.
+
+ Resolves: #1462
+
+#### res_ari: Ensure outbound websocket config has a websocket_client_id.
+ Author: George Joseph
+ Date: 2025-09-12
+
+ Added a check to outbound_websocket_apply() that makes sure an outbound
+ websocket config object in ari.conf has a websocket_client_id parameter.
+
+ Resolves: #1457
+
-<html><head><title>Readme for asterisk-22.6.0-rc1</title></head><body>
+<html><head><title>Readme for asterisk-22.6.0-rc2</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1>
<pre><code>By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
<p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
-<p><a href="ChangeLogs/ChangeLog-22.6.0-rc1.html">Change Logs</a></p>
+<p><a href="ChangeLogs/ChangeLog-22.6.0-rc2.html">Change Logs</a></p>
<!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3>
read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
-[Change Logs](ChangeLogs/ChangeLog-22.6.0-rc1.html)
+[Change Logs](ChangeLogs/ChangeLog-22.6.0-rc2.html)
<!-- END-CHANGELOGS -->
### NEW INSTALLATIONS
projects / thirdparty / asterisk.git / commitdiff
