diff options
| author | Mark Michelson <mmichelson@digium.com> | 2017-03-07 14:13:02 -0600 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2017-03-15 16:34:13 -0500 |
| commit | 10fa49e3275e81860341d5ee5e0b9d15bafd9acd (patch) | |
| tree | 5e3572e8fe93540cf2b297cf7a0665c9d329578d /UPGRADE.txt | |
| parent | e536ef7afb68751fdafd56e0e9971de3afeeffd8 (diff) | |
Add rtcp-mux support
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
Diffstat (limited to 'UPGRADE.txt')
| -rw-r--r-- | UPGRADE.txt | 20 |
1 files changed, 17 insertions, 3 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index 569cc9222..2275580ca 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -23,9 +23,23 @@ === UPGRADE-14.txt -- Upgrade info for 13 to 14 =========================================================== -Build System: - - The LOW_MEMORY compile option no longer disables inline API. To disable - inline API you must use the DISABLE_INLINE option. +From 14.3.0 to 14.4.0: + +res_rtp_asterisk: + - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP + Data and Control Packets on a Single Port." So far, the only channel driver + that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on + a PJSIP endpoint in pjsip.conf to enable the feature. + +New in 14.0.0 + +ARI: + - The policy for when to send "Dial" events has changed. Previously, "Dial" + events were sent on the calling channel's topic. However, starting in Asterisk + 14, if there is no calling channel on which to send the event, the event is + instead sent on the called channel's topic. Note that for the ARI channels + resource's dial operation, this means that the "Dial" events will always be + sent on the called channel's topic. Queue: - When reloading the members of a queue, the members added dynamically (i.e. |