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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3474/">https://reviewboard.asterisk.org/r/3474/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">A really good addition. I think this has to go in to all releases. Working on related issues in my juniper branch on spare time, to expose more information about the actual TLS session, like TLS/SSL version, crypto used etc. </pre>
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<a href="https://reviewboard.asterisk.org/r/3474/diff/1/?file=57790#file57790line21286" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_cli(a->fd, " Signaling: %s\n", cur->socket.type == SIP_TRANSPORT_TLS ? "TLS" : "non-TLS");</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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 :-)</pre>
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<p>- Olle E Johansson</p>
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<p>On April 23rd, 2014, 9:52 p.m. CEST, Patrick Laimbock wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Patrick Laimbock.</div>
<p style="color: grey;"><i>Updated April 23, 2014, 9:52 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23564">ASTERISK-23564</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/11/channels/chan_sip.c <span style="color: grey">(412921)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3474/diff/" style="margin-left: 3em;">View Diff</a></p>
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