mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
1 week ago
parent
86e3ed9620
commit
6249324f40
@ -1 +1 @@
|
|||||||
ChangeLogs/ChangeLog-23.0.0-rc1.html
|
ChangeLogs/ChangeLog-23.0.0-rc2.html
|
||||||
@ -1 +1 @@
|
|||||||
ChangeLogs/ChangeLog-23.0.0-rc1.md
|
ChangeLogs/ChangeLog-23.0.0-rc2.md
|
||||||
@ -0,0 +1,81 @@
|
|||||||
|
<html><head><title>ChangeLog for asterisk-23.0.0-rc2</title></head><body>
|
||||||
|
<h2>Change Log for Release asterisk-23.0.0-rc2</h2>
|
||||||
|
<h3>Links:</h3>
|
||||||
|
<ul>
|
||||||
|
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html">Full ChangeLog</a> </li>
|
||||||
|
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2">GitHub Diff</a> </li>
|
||||||
|
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-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>
|
||||||
@ -0,0 +1,95 @@
|
|||||||
|
|
||||||
|
## Change Log for Release asterisk-23.0.0-rc2
|
||||||
|
|
||||||
|
### Links:
|
||||||
|
|
||||||
|
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html)
|
||||||
|
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2)
|
||||||
|
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-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
|
||||||
|
|
||||||
Loading…
Reference in new issue