[asterisk-bugs] [JIRA] (ASTERISK-26378) secure_bridge_signaling and secure_bridge_media always empty

[Andrea Sannucci (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26378?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232283#comment-232283 ] 

Andrea Sannucci commented on ASTERISK-26378:
--------------------------------------------

My dialplan:

for SIP:

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

for PJSIP

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

when from BLINK Softphone with TLS and SRTP correctly configured and call a SIP extension:

Executing [1000 at externas-sip:1] NoOp("SIP/1005-00000005", "Protocolo SIPTLS = ") in new stack
    -- Executing [1000 at externas-sip:2] NoOp("SIP/1005-00000005", "Protocolo SRTP  = ") in new stack
    -- Executing [1000 at externas-sip:3] Dial("SIP/1005-00000005", "SIP/1000,45,hHkKtTwWxX") in new stack

Fron PJSIP:

 Executing [1000 at externas-pjsip:1] NoOp("PJSIP/1005-00000000", "Protocolo SIPTLS = ") in new stack
    -- Executing [1000 at externas-pjsip:2] NoOp("PJSIP/1005-00000000", "Protocolo SRTP  = ") in new stack
    -- Executing [1000 at externas-pjsip:3] Dial("PJSIP/1005-00000000", "PJSIP/1000,45,hHkKtTwWxX") in new stack

Invite from BLINK:

INVITE sip:1000 at 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 at sip11.voztovoice.org>;tag=9fc03ae80168404e9bf5995381dbcf7e
To: <sip:1000 at sip11.voztovoice.org>
Contact: <sip:94537620 at 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






> secure_bridge_signaling and secure_bridge_media always empty
> ------------------------------------------------------------
>
>                 Key: ASTERISK-26378
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26378
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Channels/chan_sip/General
>    Affects Versions: 13.11.0
>         Environment: CentOS 7.2 VPS
>            Reporter: Andrea Sannucci
>            Assignee: Andrea Sannucci
>            Severity: Minor
>
> 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



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