[asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI

[Matt Jordan]

> On April 24, 2014, 2:31 a.m., Olle E Johansson wrote:
> > /branches/11/channels/chan_sip.c, line 21286
> > <https://reviewboard.asterisk.org/r/3474/diff/1/?file=57790#file57790line21286>
> >
> >     If the header is just "signalling" we should also list other transports than TLS - UDP, TCP, WS, WSS. "non-TLS" is not a good solution :-)
> 
> Patrick Laimbock wrote:
>     Oej: thank you for your comment. I totally agree. The reason for TLS or non-TLS is my very limited C foo. I did some digging and came up with the patch below which seems to work (tested UDP/TLS,RTP/SRTP). Is that more like it?
>     
>     diff -uNr asterisk-11.9.0.org/channels/chan_sip.c asterisk-11.9.0/channels/chan_sip.c
>     --- asterisk-11.9.0.org/channels/chan_sip.c	2014-04-21 22:56:05.000000000 +0200
>     +++ asterisk-11.9.0/channels/chan_sip.c	2014-04-24 16:14:05.116999990 +0200
>     @@ -21294,6 +21294,24 @@
>       				}
>      			}
>      
>     +			/* add transport and media types */
>     +			char *transport_type;
>     +			if (cur->socket.type ==  SIP_TRANSPORT_TLS) {
>     +				transport_type = "TLS";
>     +			} else if (cur->socket.type ==  SIP_TRANSPORT_UDP) {
>     +				transport_type = "UDP";
>     +			} else if (cur->socket.type ==  SIP_TRANSPORT_TCP) {
>     +				transport_type = "TCP";
>     +			} else if (cur->socket.type ==  SIP_TRANSPORT_WS) {
>     +				transport_type = "WS";
>     +			} else if (cur->socket.type ==  SIP_TRANSPORT_WSS) {
>     +				transport_type = "WSS";
>     +			} else
>     +				transport_type = "Unknown";
>     +			
>     +			ast_cli(a->fd, "  Transport:              %s\n", transport_type);
>     +			ast_cli(a->fd, "  Media:                  %s\n", cur->srtp ? "SRTP" : "RTP");
>     +
>      			ast_cli(a->fd, "\n\n");
>      
>      			found++;

As an addition to an existing CLI command, I have no issue with this going into all branches. There's no reasonable risk of this negatively impacting users, and given recent SSL issues, it has an obvious benefit.


- Matt


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On April 24, 2014, 12:33 p.m., Patrick Laimbock wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3474/
> -----------------------------------------------------------
> 
> (Updated April 24, 2014, 12:33 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
> 
>

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