[asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
Mark Michelson
reviewboard at asterisk.org
Thu Apr 24 10:45:55 CDT 2014
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3474/#review11728
-----------------------------------------------------------
/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21501>
To address a potential corner case, I would suggest expanding this to
cur->srtp ? "SRTP" : cur->rtp ? "RTP" : "None"
- Mark Michelson
On April 23, 2014, 7:52 p.m., Patrick Laimbock wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3474/
> -----------------------------------------------------------
>
> (Updated April 23, 2014, 7:52 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-23564
> https://issues.asterisk.org/jira/browse/ASTERISK-23564
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP.
>
>
> Diffs
> -----
>
> /branches/11/channels/chan_sip.c 412921
>
> Diff: https://reviewboard.asterisk.org/r/3474/diff/
>
>
> Testing
> -------
>
> Testing was done on Asterisk-11.8.1 with TLS & RPT, TLS & SRTP, non-TLS & RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario.
>
>
> Thanks,
>
> Patrick Laimbock
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140424/4280f931/attachment.html>
More information about the asterisk-dev
mailing list