[Asterisk-code-review] chan pjsip: fix switching sending codec when asymmetric rtp ... (asterisk[13])

[Alexei Gradinari]

Alexei Gradinari has posted comments on this change. ( https://gerrit.asterisk.org/4453 )

Change subject: chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no
......................................................................


Patch Set 2:

> ASTERISK-26423 is the original issue and where people reported the
 > fix worked.

I don't know why people reported the fix worked.
May be they tested the only one direction.
I tested both directions and the issue 'one way audio'
happened only on call from 'buggy' device to 'non-buggy' device'

-- 
To view, visit https://gerrit.asterisk.org/4453
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: comment
Gerrit-Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexei Gradinari <alex2grad at gmail.com>
Gerrit-Reviewer: Alexei Gradinari <alex2grad at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-HasComments: No