Change Log for Release asterisk-21.11.0-rc2

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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