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Summary:ASTERISK-26378: secure_bridge_signaling and secure_bridge_media always empty
Reporter:Andrea Sannucci (voztovoice)Labels:
Date Opened:2016-09-14 12:17:09Date Closed:2016-09-14 15:29:49
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip Channels/chan_sip/General
Versions:13.11.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CentOS 7.2 VPSAttachments:
Description:Using the TWO CHANNELS VARIABLES in the dialplan, are always empy also when the call proceeds from a Softphone correctly configured with TSL and SRTP, that establish the call with Asterisk with TLS and SRTP

Same Behavior for SIP and PJSIP channels.

Regards
Comments:By: Asterisk Team (asteriskteam) 2016-09-14 12:17:10.308-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Richard Mudgett (rmudgett) 2016-09-14 12:42:38.249-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

{{secure_bridge_signaling}} and {{secure_bridge_media}} are *not* channel variables.  The only thing remotely like what you are talking about are values returned by the {{CHANNEL}} dialplan function accessed like this {{$\{CHANNEL(secure_bridge_signaling)}}} and {{$\{CHANNEL(secure_bridge_media)}}}

[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL
By: Andrea Sannucci (voztovoice) 2016-09-14 13:24:57.105-0500

My dialplan:

for SIP:
{noformat}
exten => _100[0-2,5],1,Noop(Protocolo SIPTLS = ${CHANNEL(secure_bridge_signaling)})
same =>  n,Noop(Protocolo SRTP  = ${CHANNEL(secure_bridge_media)})
same => n,Dial(SIP/${EXTEN},45,hHkKtTwWxX)
same => n,Gosub(voicemail,${DIALSTATUS},1(${EXTEN}))
same => n,Hangup
{noformat}
for PJSIP
{noformat}
exten => _100[0-5],1,Noop(Protocolo SIPTLS = ${CHANNEL(secure_bridge_signaling)})
same =>  n,Noop(Protocolo SRTP  = ${CHANNEL(secure_bridge_media)})
same => n,Dial(PJSIP/${EXTEN},45,hHkKtTwWxX)
same => n,Gosub(voicemail,${DIALSTATUS},1(${EXTEN}))
same => n,Hangup
{noformat}
when from BLINK Softphone with TLS and SRTP correctly configured and call a SIP extension:
{noformat}
Executing [1000@externas-sip:1] NoOp("SIP/1005-00000005", "Protocolo SIPTLS = ") in new stack
   -- Executing [1000@externas-sip:2] NoOp("SIP/1005-00000005", "Protocolo SRTP  = ") in new stack
   -- Executing [1000@externas-sip:3] Dial("SIP/1005-00000005", "SIP/1000,45,hHkKtTwWxX") in new stack
{noformat}
Fron PJSIP:
{noformat}
Executing [1000@externas-pjsip:1] NoOp("PJSIP/1005-00000000", "Protocolo SIPTLS = ") in new stack
   -- Executing [1000@externas-pjsip:2] NoOp("PJSIP/1005-00000000", "Protocolo SRTP  = ") in new stack
   -- Executing [1000@externas-pjsip:3] Dial("PJSIP/1005-00000000", "PJSIP/1000,45,hHkKtTwWxX") in new stack
{noformat}
Invite from BLINK:
{noformat}
INVITE sip:1000@sip11.voztovoice.org SIP/2.0
Via: SIP/2.0/TLS 192.168.1.14:60291;rport;branch=z9hG4bKPj502d575e46244863af2ae58e00291bed;alias
Max-Forwards: 70
From: "CursoAsterisk" <sip:1005@sip11.voztovoice.org>;tag=9fc03ae80168404e9bf5995381dbcf7e
To: <sip:1000@sip11.voztovoice.org>
Contact: <sip:94537620@192.168.1.14:60278;transport=tls>
Call-ID: bc2c2d7ec08f45d1b8a40ed405da87c6
CSeq: 20150 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 1.4.2 (Windows)
Content-Type: application/sdp
Content-Length: 689

v=0
o=- 3682848156 3682848156 IN IP4 192.168.1.14
s=Blink 1.4.2 (Windows)
t=0 0
m=audio 54362 RTP/SAVP 113 9 0 8 101
c=IN IP4 192.168.1.14
a=rtcp:54366
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ISzs0IsF8Fw+Hk6l8oAjHMwUcU+BFoKHz8SbGNlJ
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:dpdY2+KsJXUJPBaGAraolEhnGsHWjTgrdvPK80rz
a=ice-ufrag:2a4a1488
a=ice-pwd:6b4d4074
a=candidate:Hc0a8010e 1 UDP 2130706431 192.168.1.14 54362 typ host
a=candidate:Hc0a8010e 2 UDP 2130706430 192.168.1.14 54366 typ host
a=sendrecv
{noformat}

By: Richard Mudgett (rmudgett) 2016-09-14 13:55:09.652-0500

Hmm.  Only chan_iax2 and chan_sip support that mechanism.  From the code, it is not a status value to determine if the incoming call is encrypted.  For incoming calls *you* have to set the value on the channel which is then used to request that the outgoing channel of a {{Dial}} require encryption.
By: Richard Mudgett (rmudgett) 2016-09-14 14:07:03.705-0500

Here is some other information: http://lists.digium.com/pipermail/asterisk-dev/2016-September/075774.html
By: Andrea Sannucci (voztovoice) 2016-09-14 14:59:13.948-0500

OK, Thank you.

you can close this issue...

Regards