chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'

ASTERISK-23564 #close
Reported by: Patrick Laimbock
Review: https://reviewboard.asterisk.org/r/3474/

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413876 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 00ad21aff77964cdb8562b320d01fb7f11368d88..be1274b67fdd9a97b248b85468da487deff908c8 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -21163,6 +21163,24 @@ static char *complete_sip_notify(const char *line, const char *word, int pos, in
        return NULL;
 }
 
+static const char *transport2str(enum sip_transport transport)
+{
+       switch (transport) {
+       case SIP_TRANSPORT_TLS:
+               return "TLS";
+       case SIP_TRANSPORT_UDP:
+               return "UDP";
+       case SIP_TRANSPORT_TCP:
+               return "TCP";
+       case SIP_TRANSPORT_WS:
+               return "WS";
+       case SIP_TRANSPORT_WSS:
+               return "WSS";
+       }
+
+       return "Undefined";
+}
+
 /*! \brief Show details of one active dialog */
 static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
 {
@@ -21282,6 +21300,10 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a
                                }
                        }
 
+                       /* add transport and media types */
+                       ast_cli(a->fd, "  Transport:              %s\n", transport2str(cur->socket.type));
+                       ast_cli(a->fd, "  Media:                  %s\n", cur->srtp ? "SRTP" : cur->rtp ? "RTP" : "None");
+
                        ast_cli(a->fd, "\n\n");
 
                        found++;