mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-18 10:44:28 +00:00
Compare commits
5 Commits
23.0.0-rc1
...
23.0.0
| Author | SHA1 | Date | |
|---|---|---|---|
|
3bb594c9ca | ||
|
|
6249324f40 | ||
|
86e3ed9620 | ||
|
|
546cefb305 | ||
|
|
2b562c9531 |
@@ -1 +1 @@
|
||||
ChangeLogs/ChangeLog-23.0.0-rc1.html
|
||||
ChangeLogs/ChangeLog-23.0.0.html
|
||||
@@ -1 +1 @@
|
||||
ChangeLogs/ChangeLog-23.0.0-rc1.md
|
||||
ChangeLogs/ChangeLog-23.0.0.md
|
||||
@@ -1,17 +1,17 @@
|
||||
<html><head><title>ChangeLog for asterisk-23.0.0-rc1</title></head><body>
|
||||
<h2>Change Log for Release asterisk-23.0.0-rc1</h2>
|
||||
<html><head><title>ChangeLog for asterisk-23.0.0</title></head><body>
|
||||
<h2>Change Log for Release asterisk-23.0.0</h2>
|
||||
<h3>Links:</h3>
|
||||
<ul>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc1.html">Full ChangeLog</a> </li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0-rc1">GitHub Diff</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc1.tar.gz">Tarball</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html">Full ChangeLog</a> </li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0">GitHub Diff</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.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: 41</li>
|
||||
<li>Commit Authors: 13</li>
|
||||
<li>Issues Resolved: 32</li>
|
||||
<li>Commits: 45</li>
|
||||
<li>Commit Authors: 14</li>
|
||||
<li>Issues Resolved: 36</li>
|
||||
<li>Security Advisories Resolved: 1</li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j">GHSA-64qc-9x89-rx5j</a>: A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash</li>
|
||||
</ul>
|
||||
@@ -130,9 +130,10 @@
|
||||
<ul>
|
||||
<li>Alexei Gradinari: (1)</li>
|
||||
<li>Alexey Khabulyak: (1)</li>
|
||||
<li>Allan Nathanson: (1)</li>
|
||||
<li>Artem Umerov: (1)</li>
|
||||
<li>Ben Ford: (2)</li>
|
||||
<li>George Joseph: (4)</li>
|
||||
<li>George Joseph: (7)</li>
|
||||
<li>Igor Goncharovsky: (2)</li>
|
||||
<li>Joe Garlick: (1)</li>
|
||||
<li>Jose Lopes: (1)</li>
|
||||
@@ -177,6 +178,10 @@
|
||||
<li>1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled</li>
|
||||
<li>1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable</li>
|
||||
<li>1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage</li>
|
||||
<li>1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance</li>
|
||||
<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>
|
||||
<li>ASTERISK-30370: config: Template inheritance is incorrect for ast_variable_retrieve</li>
|
||||
</ul>
|
||||
<h3>Commits By Author:</h3>
|
||||
@@ -206,13 +211,16 @@
|
||||
<p>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</p>
|
||||
</li>
|
||||
<li>
|
||||
<h4>George Joseph (4):</h4>
|
||||
<h4>George Joseph (7):</h4>
|
||||
</li>
|
||||
<li>xmldoc.c: Fix rendering of CLI output.</li>
|
||||
<li>chan_websocket: Fix buffer overrun when processing TEXT websocket frames.</li>
|
||||
<li>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</li>
|
||||
<li>res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.</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>
|
||||
<p>res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.</p>
|
||||
<p>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</p>
|
||||
</li>
|
||||
<li>
|
||||
<h4>Igor Goncharovsky (2):</h4>
|
||||
@@ -278,6 +286,10 @@
|
||||
<li>logger.c: Remove deprecated/redundant configuration option.</li>
|
||||
<li>func_dialplan: Remove deprecated/redundant function.</li>
|
||||
<li>Update version for Asterisk 23</li>
|
||||
<li>config.c: fix saving of deep/wide template configurations</li>
|
||||
<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>
|
||||
<li>chan_websocket.c: Add DTMF messages</li>
|
||||
<li>app_queue.c: Add new global 'log_unpause_on_reason_change'</li>
|
||||
<li>app_waitforsilence.c: Use milliseconds to calculate timeout time</li>
|
||||
@@ -465,6 +477,54 @@
|
||||
<h4>Update version for Asterisk 23</h4>
|
||||
<p>Author: Ben Ford
|
||||
Date: 2025-08-13</p>
|
||||
<h4>config.c: fix saving of deep/wide template configurations</h4>
|
||||
<p>Author: Allan Nathanson
|
||||
Date: 2025-09-10</p>
|
||||
<p>Follow-on to #244 and #960 regarding how the ast_config_XXX APIs
|
||||
handle template inheritance.</p>
|
||||
<p>ast_config_text_file_save2() incorrectly suppressed variables if they
|
||||
matched any ancestor template. This broke deep chains (dropping values
|
||||
based on distant parents) and wide inheritance (ignoring last-wins order
|
||||
across multiple parents).</p>
|
||||
<p>The function now inspects the full template hierarchy to find the nearest
|
||||
effective parent (last occurrence wins). Earlier inherited duplicates are
|
||||
collapsed, explicit overrides are kept unless they exactly match the parent,
|
||||
and PreserveEffectiveContext avoids writing redundant lines.</p>
|
||||
<p>Resolves: #1451</p>
|
||||
<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>
|
||||
<h4>chan_websocket.c: Add DTMF messages</h4>
|
||||
<p>Author: Joe Garlick
|
||||
Date: 2025-09-04</p>
|
||||
@@ -1,18 +1,18 @@
|
||||
|
||||
## Change Log for Release asterisk-23.0.0-rc1
|
||||
## Change Log for Release asterisk-23.0.0
|
||||
|
||||
### Links:
|
||||
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc1.html)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0-rc1)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc1.tar.gz)
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
|
||||
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
|
||||
|
||||
### Summary:
|
||||
|
||||
- Commits: 41
|
||||
- Commit Authors: 13
|
||||
- Issues Resolved: 32
|
||||
- Commits: 45
|
||||
- Commit Authors: 14
|
||||
- Issues Resolved: 36
|
||||
- Security Advisories Resolved: 1
|
||||
- [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j): A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash
|
||||
|
||||
@@ -115,9 +115,10 @@
|
||||
|
||||
- Alexei Gradinari: (1)
|
||||
- Alexey Khabulyak: (1)
|
||||
- Allan Nathanson: (1)
|
||||
- Artem Umerov: (1)
|
||||
- Ben Ford: (2)
|
||||
- George Joseph: (4)
|
||||
- George Joseph: (7)
|
||||
- Igor Goncharovsky: (2)
|
||||
- Joe Garlick: (1)
|
||||
- Jose Lopes: (1)
|
||||
@@ -163,6 +164,10 @@
|
||||
- 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
|
||||
- 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable
|
||||
- 1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage
|
||||
- 1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance
|
||||
- 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
|
||||
- ASTERISK-30370: config: Template inheritance is incorrect for ast_variable_retrieve
|
||||
|
||||
### Commits By Author:
|
||||
@@ -179,11 +184,14 @@
|
||||
- #### Ben Ford (1):
|
||||
- res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
|
||||
|
||||
- #### George Joseph (4):
|
||||
- #### George Joseph (7):
|
||||
- xmldoc.c: Fix rendering of CLI output.
|
||||
- chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
|
||||
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
|
||||
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
|
||||
- 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.
|
||||
|
||||
- #### Igor Goncharovsky (2):
|
||||
- app_waitforsilence.c: Use milliseconds to calculate timeout time
|
||||
@@ -233,6 +241,10 @@
|
||||
- logger.c: Remove deprecated/redundant configuration option.
|
||||
- func_dialplan: Remove deprecated/redundant function.
|
||||
- Update version for Asterisk 23
|
||||
- config.c: fix saving of deep/wide template configurations
|
||||
- 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.
|
||||
- chan_websocket.c: Add DTMF messages
|
||||
- app_queue.c: Add new global 'log_unpause_on_reason_change'
|
||||
- app_waitforsilence.c: Use milliseconds to calculate timeout time
|
||||
@@ -481,6 +493,66 @@
|
||||
Date: 2025-08-13
|
||||
|
||||
|
||||
#### config.c: fix saving of deep/wide template configurations
|
||||
Author: Allan Nathanson
|
||||
Date: 2025-09-10
|
||||
|
||||
Follow-on to #244 and #960 regarding how the ast_config_XXX APIs
|
||||
handle template inheritance.
|
||||
|
||||
ast_config_text_file_save2() incorrectly suppressed variables if they
|
||||
matched any ancestor template. This broke deep chains (dropping values
|
||||
based on distant parents) and wide inheritance (ignoring last-wins order
|
||||
across multiple parents).
|
||||
|
||||
The function now inspects the full template hierarchy to find the nearest
|
||||
effective parent (last occurrence wins). Earlier inherited duplicates are
|
||||
collapsed, explicit overrides are kept unless they exactly match the parent,
|
||||
and PreserveEffectiveContext avoids writing redundant lines.
|
||||
|
||||
Resolves: #1451
|
||||
|
||||
#### 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
|
||||
|
||||
#### chan_websocket.c: Add DTMF messages
|
||||
Author: Joe Garlick
|
||||
Date: 2025-09-04
|
||||
@@ -1,4 +1,4 @@
|
||||
<html><head><title>Readme for asterisk-23.0.0-rc1</title></head><body>
|
||||
<html><head><title>Readme for asterisk-23.0.0</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.
|
||||
@@ -37,7 +37,7 @@ hardware.</p>
|
||||
<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-23.0.0-rc1.html">Change Logs</a></p>
|
||||
<p><a href="ChangeLogs/ChangeLog-23.0.0.html">Change Logs</a></p>
|
||||
<!-- END-CHANGELOGS -->
|
||||
|
||||
<h3>NEW INSTALLATIONS</h3>
|
||||
|
||||
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
|
||||
read the Change Logs.
|
||||
|
||||
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
|
||||
[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc1.html)
|
||||
[Change Logs](ChangeLogs/ChangeLog-23.0.0.html)
|
||||
<!-- END-CHANGELOGS -->
|
||||
|
||||
### NEW INSTALLATIONS
|
||||
|
||||
@@ -59,6 +59,18 @@
|
||||
/channels/answer ARI endpoint.
|
||||
</para>
|
||||
</enum>
|
||||
<enum name="p - Passthrough mode">
|
||||
<para>In passthrough mode, the channel driver won't 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.
|
||||
</para>
|
||||
</enum>
|
||||
<enum name="v(uri_parameters) - Add parameters to the outbound URI">
|
||||
<para>This option allows you to add additional parameters to the
|
||||
outbound URI. The format is:
|
||||
@@ -78,6 +90,9 @@
|
||||
<example title="Make an outbound WebSocket connection using connection 'connection1' and the 'sln16' codec.">
|
||||
same => n,Dial(WebSocket/connection1/c(sln16))
|
||||
</example>
|
||||
<example title="Make an outbound WebSocket connection using connection 'connection1' and the 'opus' codec. Passthrough mode will automatically be set.">
|
||||
same => n,Dial(WebSocket/connection1/c(opus))
|
||||
</example>
|
||||
<example title="Listen for an incoming WebSocket connection and don't auto-answer it.">
|
||||
same => n,Dial(WebSocket/INCOMING/n)
|
||||
</example>
|
||||
@@ -127,6 +142,7 @@ struct websocket_pvt {
|
||||
char *uri_params;
|
||||
char *leftover_data;
|
||||
int no_auto_answer;
|
||||
int passthrough;
|
||||
int optimal_frame_size;
|
||||
int bulk_media_in_progress;
|
||||
int report_queue_drained;
|
||||
@@ -188,6 +204,7 @@ static void set_channel_format(struct websocket_pvt * instance,
|
||||
if (ast_format_cmp(ast_channel_rawreadformat(instance->channel), fmt)
|
||||
== AST_FORMAT_CMP_NOT_EQUAL) {
|
||||
ast_channel_set_rawreadformat(instance->channel, fmt);
|
||||
ast_set_read_format(instance->channel, ast_channel_readformat(instance->channel));
|
||||
ast_debug(4, "Switching readformat to %s\n", ast_format_get_name(fmt));
|
||||
}
|
||||
}
|
||||
@@ -332,10 +349,10 @@ static struct ast_frame *webchan_read(struct ast_channel *ast)
|
||||
}
|
||||
|
||||
/*
|
||||
* If the frame length is already optimal_frame_size, we can just
|
||||
* return it.
|
||||
* If we're in passthrough mode or the frame length is already optimal_frame_size,
|
||||
* we can just return it.
|
||||
*/
|
||||
if (native_frame->datalen == instance->optimal_frame_size) {
|
||||
if (instance->passthrough || native_frame->datalen == instance->optimal_frame_size) {
|
||||
set_channel_format(instance, instance->native_format);
|
||||
return native_frame;
|
||||
}
|
||||
@@ -499,6 +516,11 @@ static int process_text_message(struct websocket_pvt *instance,
|
||||
ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
|
||||
|
||||
} else if (ast_strings_equal(command, START_MEDIA_BUFFERING)) {
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
AST_LIST_LOCK(&instance->frame_queue);
|
||||
instance->bulk_media_in_progress = 1;
|
||||
AST_LIST_UNLOCK(&instance->frame_queue);
|
||||
@@ -511,6 +533,12 @@ static int process_text_message(struct websocket_pvt *instance,
|
||||
|
||||
id = ast_strip(command + strlen(STOP_MEDIA_BUFFERING));
|
||||
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
|
||||
ast_debug(4, "%s: WebSocket %s '%s' with %d bytes in leftover_data.\n",
|
||||
ast_channel_name(instance->channel), STOP_MEDIA_BUFFERING, id,
|
||||
(int)instance->leftover_len);
|
||||
@@ -533,6 +561,13 @@ static int process_text_message(struct websocket_pvt *instance,
|
||||
|
||||
} else if (ast_strings_equal(command, FLUSH_MEDIA)) {
|
||||
struct ast_frame *frame = NULL;
|
||||
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_LIST_LOCK(&instance->frame_queue);
|
||||
while ((frame = AST_LIST_REMOVE_HEAD(&instance->frame_queue, frame_list))) {
|
||||
ast_frfree(frame);
|
||||
@@ -543,6 +578,12 @@ static int process_text_message(struct websocket_pvt *instance,
|
||||
AST_LIST_UNLOCK(&instance->frame_queue);
|
||||
|
||||
} else if (ast_strings_equal(payload, REPORT_QUEUE_DRAINED)) {
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_LIST_LOCK(&instance->frame_queue);
|
||||
instance->report_queue_drained = 1;
|
||||
AST_LIST_UNLOCK(&instance->frame_queue);
|
||||
@@ -569,11 +610,21 @@ static int process_text_message(struct websocket_pvt *instance,
|
||||
}
|
||||
|
||||
} else if (ast_strings_equal(payload, PAUSE_MEDIA)) {
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
AST_LIST_LOCK(&instance->frame_queue);
|
||||
instance->queue_paused = 1;
|
||||
AST_LIST_UNLOCK(&instance->frame_queue);
|
||||
|
||||
} else if (ast_strings_equal(payload, CONTINUE_MEDIA)) {
|
||||
if (instance->passthrough) {
|
||||
ast_debug(4, "%s: WebSocket in passthrough mode. Ignoring %s command.\n",
|
||||
ast_channel_name(instance->channel), command);
|
||||
return 0;
|
||||
}
|
||||
AST_LIST_LOCK(&instance->frame_queue);
|
||||
instance->queue_paused = 0;
|
||||
AST_LIST_UNLOCK(&instance->frame_queue);
|
||||
@@ -607,6 +658,11 @@ static int process_binary_message(struct websocket_pvt *instance,
|
||||
next_frame_ptr = payload;
|
||||
instance->bytes_read += payload_len;
|
||||
|
||||
if (instance->passthrough) {
|
||||
res = queue_frame_from_buffer(instance, payload, payload_len);
|
||||
return res;
|
||||
}
|
||||
|
||||
if (instance->bulk_media_in_progress && instance->leftover_len > 0) {
|
||||
/*
|
||||
* We have leftover data from a previous websocket message.
|
||||
@@ -791,10 +847,10 @@ static void *read_thread_handler(void *obj)
|
||||
* This is especially important for outbound connections otherwise
|
||||
* the app won't know who the media is for.
|
||||
*/
|
||||
res = ast_asprintf(&command, "%s connection_id:%s channel:%s format:%s optimal_frame_size:%d", MEDIA_START,
|
||||
res = ast_asprintf(&command, "%s connection_id:%s channel:%s format:%s optimal_frame_size:%d ptime:%d", MEDIA_START,
|
||||
instance->connection_id, ast_channel_name(instance->channel),
|
||||
ast_format_get_name(instance->native_format),
|
||||
instance->optimal_frame_size);
|
||||
instance->optimal_frame_size, instance->native_codec->default_ms);
|
||||
if (res <= 0 || !command) {
|
||||
ast_queue_control(instance->channel, AST_CONTROL_HANGUP);
|
||||
ast_log(LOG_ERROR, "%s: Failed to create MEDIA_START\n", ast_channel_name(instance->channel));
|
||||
@@ -843,9 +899,11 @@ static int webchan_write(struct ast_channel *ast, struct ast_frame *f)
|
||||
ast_channel_name(ast));
|
||||
return -1;
|
||||
}
|
||||
if (f->subclass.format != instance->native_format) {
|
||||
ast_log(LOG_WARNING, "%s: This WebSocket channel only supports the '%s' format\n",
|
||||
ast_channel_name(ast), ast_format_get_name(instance->native_format));
|
||||
|
||||
if (ast_format_cmp(f->subclass.format, instance->native_format) == AST_FORMAT_CMP_NOT_EQUAL) {
|
||||
ast_log(LOG_WARNING, "%s: This WebSocket channel only supports the '%s' format, not '%s'\n",
|
||||
ast_channel_name(ast), ast_format_get_name(instance->native_format),
|
||||
ast_format_get_name(f->subclass.format));
|
||||
return -1;
|
||||
}
|
||||
|
||||
@@ -1044,15 +1102,36 @@ static struct websocket_pvt* websocket_new(const char *chan_name,
|
||||
* References for native_format and native_codec are now held by the
|
||||
* instance and will be released when the instance is destroyed.
|
||||
*/
|
||||
instance->optimal_frame_size =
|
||||
(instance->native_codec->default_ms * instance->native_codec->minimum_bytes)
|
||||
/ instance->native_codec->minimum_ms;
|
||||
|
||||
instance->leftover_data = ast_calloc(1, instance->optimal_frame_size);
|
||||
if (!instance->leftover_data) {
|
||||
return NULL;
|
||||
/*
|
||||
* It's not possible for us to re-time or re-frame media if the data
|
||||
* stream can't be broken up on arbitrary byte boundaries. This is usually
|
||||
* indicated by the codec's minimum_bytes being small (10 bytes or less).
|
||||
* We need to force passthrough mode in this case.
|
||||
*/
|
||||
if (instance->native_codec->minimum_bytes <= 10) {
|
||||
instance->passthrough = 1;
|
||||
instance->optimal_frame_size = 0;
|
||||
} else {
|
||||
instance->optimal_frame_size =
|
||||
(instance->native_codec->default_ms * instance->native_codec->minimum_bytes)
|
||||
/ instance->native_codec->minimum_ms;
|
||||
instance->leftover_data = ast_calloc(1, instance->optimal_frame_size);
|
||||
if (!instance->leftover_data) {
|
||||
return NULL;
|
||||
}
|
||||
}
|
||||
|
||||
ast_debug(3,
|
||||
"%s: WebSocket channel native format '%s' Sample rate: %d ptime: %dms minms: %u minbytes: %u passthrough: %d optimal_frame_size: %d\n",
|
||||
chan_name, ast_format_get_name(instance->native_format),
|
||||
ast_format_get_sample_rate(instance->native_format),
|
||||
ast_format_get_default_ms(instance->native_format),
|
||||
ast_format_get_minimum_ms(instance->native_format),
|
||||
ast_format_get_minimum_bytes(instance->native_format),
|
||||
instance->passthrough,
|
||||
instance->optimal_frame_size);
|
||||
|
||||
/* We have exclusive access to proxy and sorcery, no need for locking here. */
|
||||
if (ao2_weakproxy_set_object(proxy, instance, OBJ_NOLOCK)) {
|
||||
return NULL;
|
||||
@@ -1195,12 +1274,14 @@ enum {
|
||||
OPT_WS_CODEC = (1 << 0),
|
||||
OPT_WS_NO_AUTO_ANSWER = (1 << 1),
|
||||
OPT_WS_URI_PARAM = (1 << 2),
|
||||
OPT_WS_PASSTHROUGH = (1 << 3),
|
||||
};
|
||||
|
||||
enum {
|
||||
OPT_ARG_WS_CODEC,
|
||||
OPT_ARG_WS_NO_AUTO_ANSWER,
|
||||
OPT_ARG_WS_URI_PARAM,
|
||||
OPT_ARG_WS_PASSTHROUGH,
|
||||
OPT_ARG_ARRAY_SIZE
|
||||
};
|
||||
|
||||
@@ -1208,6 +1289,7 @@ AST_APP_OPTIONS(websocket_options, BEGIN_OPTIONS
|
||||
AST_APP_OPTION_ARG('c', OPT_WS_CODEC, OPT_ARG_WS_CODEC),
|
||||
AST_APP_OPTION('n', OPT_WS_NO_AUTO_ANSWER),
|
||||
AST_APP_OPTION_ARG('v', OPT_WS_URI_PARAM, OPT_ARG_WS_URI_PARAM),
|
||||
AST_APP_OPTION('p', OPT_WS_PASSTHROUGH),
|
||||
END_OPTIONS );
|
||||
|
||||
static struct ast_channel *webchan_request(const char *type,
|
||||
@@ -1281,6 +1363,9 @@ static struct ast_channel *webchan_request(const char *type,
|
||||
}
|
||||
|
||||
instance->no_auto_answer = ast_test_flag(&opts, OPT_WS_NO_AUTO_ANSWER);
|
||||
if (!instance->passthrough) {
|
||||
instance->passthrough = ast_test_flag(&opts, OPT_WS_PASSTHROUGH);
|
||||
}
|
||||
|
||||
if (ast_test_flag(&opts, OPT_WS_URI_PARAM)
|
||||
&& !ast_strlen_zero(opt_args[OPT_ARG_WS_URI_PARAM])) {
|
||||
|
||||
@@ -119,6 +119,11 @@ static int outbound_websocket_apply(const struct ast_sorcery *sorcery, void *obj
|
||||
|
||||
ast_debug(3, "%s: Initializing outbound websocket\n", id);
|
||||
|
||||
if (!owc->websocket_client) {
|
||||
ast_log(LOG_WARNING, "%s: Outbound websocket missing websocket_client_id\n", id);
|
||||
res = -1;
|
||||
}
|
||||
|
||||
if (ast_strlen_zero(owc->apps)) {
|
||||
ast_log(LOG_WARNING, "%s: Outbound websocket missing apps\n", id);
|
||||
res = -1;
|
||||
|
||||
@@ -3453,7 +3453,7 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
|
||||
int res;
|
||||
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
|
||||
char *out = buf;
|
||||
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
||||
struct dtls_details *dtls = (!rtcp || rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_MUX) ? &rtp->dtls : &rtp->rtcp->dtls;
|
||||
|
||||
/* Don't send RTP if DTLS hasn't finished yet */
|
||||
if (dtls->ssl && ((*out < 20) || (*out > 63)) && dtls->connection == AST_RTP_DTLS_CONNECTION_NEW) {
|
||||
|
||||
Reference in New Issue
Block a user