mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
1 week ago
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ChangeLogs/ChangeLog-23.0.0-rc1.html
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ChangeLogs/ChangeLog-23.0.0-rc2.html
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ChangeLogs/ChangeLog-23.0.0-rc1.md
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ChangeLogs/ChangeLog-23.0.0-rc2.md
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<html><head><title>ChangeLog for asterisk-23.0.0-rc2</title></head><body>
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<h2>Change Log for Release asterisk-23.0.0-rc2</h2>
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<h3>Links:</h3>
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<ul>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html">Full ChangeLog</a> </li>
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<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2">GitHub Diff</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc2.tar.gz">Tarball</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
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</ul>
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<h3>Summary:</h3>
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<ul>
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<li>Commits: 3</li>
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<li>Commit Authors: 1</li>
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<li>Issues Resolved: 3</li>
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<li>Security Advisories Resolved: 0</li>
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</ul>
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<h3>User Notes:</h3>
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<h3>Upgrade Notes:</h3>
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<h3>Developer Notes:</h3>
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<h3>Commit Authors:</h3>
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<ul>
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<li>George Joseph: (3)</li>
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</ul>
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<h2>Issue and Commit Detail:</h2>
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<h3>Closed Issues:</h3>
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<ul>
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<li>1457: [bug]: segmentation fault because of a wrong ari config</li>
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<li>1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.</li>
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<li>1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes</li>
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</ul>
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<h3>Commits By Author:</h3>
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<ul>
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<li>
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<h4>George Joseph (3):</h4>
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</li>
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<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
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<li>chan_websocket: Fix codec validation and add passthrough option.</li>
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<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
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</ul>
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<h3>Commit List:</h3>
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<ul>
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<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
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<li>chan_websocket: Fix codec validation and add passthrough option.</li>
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<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
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</ul>
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<h3>Commit Details:</h3>
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<h4>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</h4>
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<p>Author: George Joseph
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Date: 2025-09-23</p>
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<p>In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
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needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
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AST_RTP_INSTANCE_RTCP_MUX is set.</p>
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<p>Resolves: #1474</p>
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<h4>chan_websocket: Fix codec validation and add passthrough option.</h4>
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<p>Author: George Joseph
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Date: 2025-09-17</p>
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<ul>
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<li>Fixed an issue in webchan_write() where we weren't detecting equivalent
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codecs properly.</li>
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<li>Added the "p" dialstring option that puts the channel driver in
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"passthrough" mode where it will not attempt to re-frame or re-time
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media coming in over the websocket from the remote app. This can be used
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for any codec but MUST be used for codecs that use packet headers or whose
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data stream can't be broken up on arbitrary byte boundaries. In this case,
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the remote app is fully responsible for correctly framing and timing media
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sent to Asterisk and the MEDIA text commands that could be sent over the
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websocket are disabled. Currently, passthrough mode is automatically set
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for the opus, speex and g729 codecs.</li>
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<li>Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
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ensure proper translation paths are set up when switching between native
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frames and slin silence frames. This fixes an issue with codec errors
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when transcode_via_sln=yes.</li>
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</ul>
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<p>Resolves: #1462</p>
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<h4>res_ari: Ensure outbound websocket config has a websocket_client_id.</h4>
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<p>Author: George Joseph
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Date: 2025-09-12</p>
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<p>Added a check to outbound_websocket_apply() that makes sure an outbound
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websocket config object in ari.conf has a websocket_client_id parameter.</p>
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<p>Resolves: #1457</p>
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</body></html>
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## Change Log for Release asterisk-23.0.0-rc2
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### Links:
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- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc2.tar.gz)
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- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
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### Summary:
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- Commits: 3
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- Commit Authors: 1
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- Issues Resolved: 3
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- Security Advisories Resolved: 0
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### User Notes:
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### Upgrade Notes:
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### Developer Notes:
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### Commit Authors:
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- George Joseph: (3)
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## Issue and Commit Detail:
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### Closed Issues:
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- 1457: [bug]: segmentation fault because of a wrong ari config
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- 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
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- 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes
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### Commits By Author:
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- #### George Joseph (3):
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- res_ari: Ensure outbound websocket config has a websocket_client_id.
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- chan_websocket: Fix codec validation and add passthrough option.
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- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
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### Commit List:
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- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
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- chan_websocket: Fix codec validation and add passthrough option.
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- res_ari: Ensure outbound websocket config has a websocket_client_id.
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### Commit Details:
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#### res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
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Author: George Joseph
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Date: 2025-09-23
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In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
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needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
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AST_RTP_INSTANCE_RTCP_MUX is set.
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Resolves: #1474
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#### chan_websocket: Fix codec validation and add passthrough option.
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Author: George Joseph
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Date: 2025-09-17
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* Fixed an issue in webchan_write() where we weren't detecting equivalent
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codecs properly.
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* Added the "p" dialstring option that puts the channel driver in
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"passthrough" mode where it will not attempt to re-frame or re-time
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media coming in over the websocket from the remote app. This can be used
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for any codec but MUST be used for codecs that use packet headers or whose
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data stream can't be broken up on arbitrary byte boundaries. In this case,
|
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the remote app is fully responsible for correctly framing and timing media
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sent to Asterisk and the MEDIA text commands that could be sent over the
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websocket are disabled. Currently, passthrough mode is automatically set
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for the opus, speex and g729 codecs.
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* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
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ensure proper translation paths are set up when switching between native
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frames and slin silence frames. This fixes an issue with codec errors
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when transcode_via_sln=yes.
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Resolves: #1462
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#### res_ari: Ensure outbound websocket config has a websocket_client_id.
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Author: George Joseph
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Date: 2025-09-12
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Added a check to outbound_websocket_apply() that makes sure an outbound
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websocket config object in ari.conf has a websocket_client_id parameter.
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Resolves: #1457
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