> On April 24, 2014, 7:31 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 :-)
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++;
- Patrick
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3474/#review11726
-----------------------------------------------------------
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
>
>
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