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
+Author: George Joseph + Date: 2025-09-17
+Resolves: #1462
+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
+ diff --git a/ChangeLogs/ChangeLog-23.0.0-rc2.md b/ChangeLogs/ChangeLog-23.0.0-rc2.md new file mode 100644 index 0000000000..1feb3732f9 --- /dev/null +++ b/ChangeLogs/ChangeLog-23.0.0-rc2.md @@ -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 + diff --git a/README.html b/README.html index 48dbe2cb60..35bf6fe948 100644 --- a/README.html +++ b/README.html @@ -1,4 +1,4 @@ -By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.
If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
-
+
NEW INSTALLATIONS
diff --git a/README.md b/README.md
index eefe89c9d8..92421abf1b 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
-[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc1.html)
+[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc2.html)
### NEW INSTALLATIONS