mirror of
https://github.com/asterisk/asterisk.git
synced 2026-07-23 07:54:50 -07:00
Mark Michelson
a7764b0c36
This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
102 lines
4.5 KiB
Plaintext
102 lines
4.5 KiB
Plaintext
===========================================================
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===
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=== Information for upgrading between Asterisk versions
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===
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also include advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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=== UPGRADE-13.txt -- Upgrade info for 12 to 13
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===========================================================
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From 14.3.0 to 14.4.0:
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res_rtp_asterisk:
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- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
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Data and Control Packets on a Single Port." So far, the only channel driver
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that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
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a PJSIP endpoint in pjsip.conf to enable the feature.
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New in 14.0.0
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ARI:
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- The policy for when to send "Dial" events has changed. Previously, "Dial"
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events were sent on the calling channel's topic. However, starting in Asterisk
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14, if there is no calling channel on which to send the event, the event is
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instead sent on the called channel's topic. Note that for the ARI channels
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resource's dial operation, this means that the "Dial" events will always be
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sent on the called channel's topic.
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Channel Drivers:
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chan_dahdi:
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- For users using the FXO port (FXS signaling) distinctive ring detection
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feature, you will need to adjust the dringX count values. The count
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values now only record ring end events instead of any DAHDI event. A
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ring-ring-ring pattern would exceed the pattern limits and stop
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Caller-ID detection.
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chan_sip:
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- The SIP dial string has been extended past the [!dnid] option by another
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exclamation mark: [!dnid[!fromuri]. An exclamation mark in the To-URI
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will now mean changes to the From-URI.
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Core:
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- The REF_DEBUG compiler flag is now used to enable refdebug by default.
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The setting can be overridden in asterisk.conf by setting refdebug in
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the options category. No recompile is required to enable/disable it.
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- Modified processing of command-line options to first parse only what
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is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
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the remaining options are processed. The -X option now applies to
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asterisk.conf only. To enable #exec for other config files you must
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set execincludes=yes in asterisk.conf. Any other option set on the
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command-line will now override the equivalent setting from asterisk.conf.
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AMI:
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- The 'ModuleCheck' Action's Version key will no longer show the module
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version. The value will always be blank.
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CLI:
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- The 'core show file version' command has been removed. When Asterisk
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moved to Git, the source control version support was removed. As a
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result, the CLi command was no longer useful and was removed as well.
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Logging:
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- The first callid created is now 1 instead of 0. The value 0
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is now reserved to represent a lack of callid.
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AMI:
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- The Command action now sends the output from the CLI command as a series
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of Output headers for each line instead of as a block of text with the
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--END COMMAND-- delimiter to match the output from other actions.
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Commands that fail to execute (no such command, invalid syntax etc.) now
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return an Error response instead of Success.
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app_amd:
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- The 'maximum_number_of_words' configuration option and parameter to the AMD
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application previously did not match the documented functionality + variable
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name. In Asterisk 13, a value of '3' would mean that if '3' words were detected,
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the result would be detection as a 'MACHINE'. As of this version, the value
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reflects the maximum words that if EXCEEDED (rather than reached), would
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result in detection as a machine. This means that you should update this
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value to be one higher than your previos value, if your previous value
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was working well for you.
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===========================================================
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===========================================================
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