[Asterisk-code-review] res_rtp_asterisk: fix memory leak (asterisk[master])

Jean Aunis - Prescom asteriskteam at digium.com
Wed Sep 29 10:42:59 CDT 2021


Attention is currently required from: Sean Bright, Joshua Colp.
Jean Aunis - Prescom has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/16561 )

Change subject: res_rtp_asterisk: fix memory leak
......................................................................


Patch Set 2:

(1 comment)

Patchset:

PS1: 
> This appears to be correct. […]
It's a simple outbound call with chan_sip. But I don't see either how it is possible no to call ast_rtp_ice_start... there must be something unusual in the SIP response we receive. I need to check that.



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Gerrit-Project: asterisk
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