]> git.ipfire.org Git - thirdparty/asterisk.git/commit
chan_sip: Filter pass-through audio/video formats away, again.
authorAlexander Traud <pabstraud@compuserve.com>
Fri, 5 Feb 2021 12:29:51 +0000 (13:29 +0100)
committerGeorge Joseph <gjoseph@digium.com>
Tue, 23 Feb 2021 18:41:15 +0000 (12:41 -0600)
commit5894535fedcd129ff33459a5cbdd3acf0f3304bd
tree25c2b98843b2d6e6d3a0a319a81eb2cf9459a40e
parentb0f349a330856c19b1f54b5229bddddabc5b9d2b
chan_sip: Filter pass-through audio/video formats away, again.

Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.

This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.

Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34
channels/chan_sip.c
main/translate.c