[asterisk-bugs] [JIRA] (ASTERISK-20461) channel originate Local/foo * forces translation via slin

[lvl (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20461?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=250100#comment-250100 ] 

lvl commented on ASTERISK-20461:
--------------------------------

As Joshua found, I created a ticket that's pretty much a duplicate of this case in ASTERISK-28785. For anyone interested, more information can be found there.

>  at which point once both sides have answered, it would optimize itself away and the two sides - which have the same format - would be made compatible and no transcoding would occur.

An interesting fact that wasn't mentioned in this ticket yet, is that this does in fact work properly _sometimes_ in _some Asterisk versions_.

I have a reproduction testcase that automatically detects whether the unnecessary transcoding is performed (manager originate without any codecs, both SIP clients only configured with g722):

In Asterisk 15.3 it fails pretty much 50% of the time

{code}
4 fails, 6 succeeds
{code}

In Asterisk 16.3 and 16.8 it fails 100% of the time

{code}
10 fails, 0 succeeds
{code}

In Asterisk 16.3 with commit https://github.com/asterisk/asterisk/commit/a46fcaca7bcb019a1a2b29d3c9a528da637840d7 reverted it's back to only failing about 50% of the time

{code}
4 fails, 6 succeeds
{code}

I have no clue why commit a46fcaca7bcb019a1a2b29d3c9a528da637840d7 is a culprit here, I found it using bisect. I guess it subtly changes the timing which causes the bridge reconfiguration to fail.

> channel originate Local/foo * forces translation via slin
> ---------------------------------------------------------
>
>                 Key: ASTERISK-20461
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20461
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Applications/app_originate, Channels/chan_local
>    Affects Versions: 1.8.12.2, 1.8.15.1, 13.18.4
>            Reporter: Jaco Kroon
>         Attachments: asterisk-chan_local_g729.patch, asterisk-originate-with-all.patch, asterisk-originate-with-g729.patch, slin.txt, trace.txt
>
>
> The simplest example of this is with a SIP and an IAX2 endpoint available, eg SIP/102 and IAX2/upstream, both configured with disallow=all; allow=g729 with codecs from digium loaded.
> cli> originate Local/102 at aanswer Dial IAX2/upstream/number
> once connected a g729 show licenses will show that two encoders and two decoders are in use.



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