Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated May 13, 2014, 12:40 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 413876 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 25, 2014, 5:37 p.m.) Review request for Asterisk Developers. Changes --- Updated the patch to remove the red blob, put declaration of transport_type at the top and add the curlies, all per rmudgett. 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 (updated) - /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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
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 22:56:05.0
+0200
+++ asterisk-11.9.0/channels/chan_sip.c 2014-04-24 16:14:05.11690 +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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- (Updated April 24, 2014, 5:33 p.m.) Review request for Asterisk Developers. Changes --- Added all transport types (UDP, TCP, TLS, WS, WSS) as suggested by oej and added corner case as suggested by mmichelson. Tested UDP/RTP TLS/SRTP on Asterisk-11.9.0. 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 (updated) - /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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/ --- Review request for Asterisk Developers. 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI
--- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
