Displaying 20 results from an estimated 151 matches for "dtls".
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2007 Jul 09
2
DTLS for Centos?
Is DTLS available for Centos? Either Centos 4 or 5.
DTLS is TLS over UDP. Highly valued to protect SIP traffic.....
2014 May 20
1
How to enable DTLS
Hi All,
Currently i am integrating webRTC demo.
I have issue using firefox,someone suggest me to enable DTLS for webRTC
working in firefox using Asterisk.
I am using Asterisk 11.9.0.
https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J
Can any one tell me how to enable DTLS ?
--
Thanks,
Bhavik Patel
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2015 Oct 28
2
Receiving Messages and Extensions Config for WebRTC
...ing
directmedia=no ; Asterisk will relay media for this peer
transport=ws ; Asterisk will allow this peer to register on UDP or
WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
nat=force_rport,comedia
accept_outofcall_message=yes
outofcall_message_context=messages
;dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
;dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS cert file is
;dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS private key...
2014 Jul 07
0
no audio on call from sipML5 in browsers to Asterisk 11 with DTLS-SRTP
Hi all !
I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in
order to test WebRTC setup on my Asterisk PBX. I am using latest SVN
version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677)
If I make calls from softphones (Zoiper, X-Lite), which do not support
DTLS at all, I can hear the Echo Test sound.
BUT when I call from browser (I've tried latest Mozilla Firefox and
Chrome on both Windows 8.1 x86 and Android 4.3), I can successfully
establish a call but I can hear NO SOUND.
As it turns out (according to wireshark logs), Asterick is sending me
DT...
2015 Aug 12
2
webrtc no audio
Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
> Vinicius Fontes wrote:
>> I'm having the same issue! The difference in my case is Asterisk server
>> has a public IPv4 and the browser is behind a single NAT.
>>
>> I'm forwarding my configuration below (which I posted previously on
>> asterisk-users).
>>
>> How can we debug ICE negotiation?
>
2015 Jun 16
1
Req help regarding webRTC : Attempted Attempted to set an invalid DTLS-SRTP configuration on RTP instance
...ng to setup a Asterisk setup in AWS instance Centos6.5 . I
have installed Asterisk 13.4 with srtp,pjproject. I have configured two
numbers for webRTC clients, when i try to call from a client (sipml5) to
another client (sipml5) it throws the following error:
"chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid
DTLS-SRTP configuration on RTP instance '0x7f3ccc020718'"
I am struck here.
Please throw some light to go further.
Thanks in advance.
Best regards,
Ruban.S
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2013 May 17
1
DTLS
Hi all,
I am looking for a secured communication between web clients and my
servers. tinc looks great. I understand it uses UDP for data. But does
it use DTLS (newbbie question) ?
As someone tryed to use 0MQ with it ?
Cheers,
Laurent.
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2014 Jan 28
0
DTLS setting impacts encryption setting
...hout encryption, but
will be happy to accept incoming calls regardless of whether the caller
wants encryption or not
If encryption=yes, then Asterisk not only uses encryption for the
outgoing calls but it will refuse to accept incoming calls unless they
use encryption too
If I have
encryption=no
dtlsenable=yes
the DTLS support works but Asterisk will no longer accept incoming calls
using regular RTP/AVP. These messages appear in the console and the
call is rejected with code 488:
[Jan 28 11:08:42] WARNING[24673][C-00000009]: chan_sip.c:10496
process_sdp: Processed DTLS [FALSE]
[Jan 28 11:08:...
2014 Apr 25
1
srtp/dtls when sip is clear over lo
Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or
chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly,
will ast negotiate srtp or dtls even ast and the proxy speak sip in
the clear over the lo interface?
Avoiding encryption over lo can aid debugging, but will doing so also
block secure media?
-JimC
--
James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
2016 Oct 05
2
Ast 13.10 to 13.11 stop working webrtc
...g, failing with
chan_sip.c:4083 retrans_pkt: Hanging up call
7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
is there any way to configure to have the previous behaviour?
Im trying to set dtlscipher=AES128-SHA but I always see
DTLS ECDH initialized (automatic), faster PFS enabled
any idea?
Thanks!
res_rtp_asterisk
------------------
* The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
Enabling PFS is attempted by default, and is dependent on the configuration
of the...
2007 Jul 09
2
Background transfers with callback
Hello list,
I have successfully set up Asterisk, but girls from our office complain
to me that when they hit Flash to transfer a call and pick the number,
they need to wait until the call is answered, and only then they could
hangup.
On the analog PBX we had before the transfer was in "background", and
if called party did not answer the call, then the call returned to
the girl in the
2015 Jul 27
0
Asterisk 11.19.0-rc1 Now Available
...registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handsh...
2015 Aug 07
0
Asterisk 11.19.0 Now Available
...registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handsh...
2013 Jun 17
1
Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
...nes:
Rejecting secure audio stream without encryption details: audio 62583
UDP/TLS/RTP/SAVPF 109 0 8 101
Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060
No compatible codecs for this SIP call.
Here is the sip.conf info. I have tried various permutations of the dtls
and encryption parameters with no luck. I do have openssl and srtp built
into Asterisk (that solved a different error dealing with the RTP engine).
[webrtc-dtls] ; Add DTLS stuff for Mozilla Nightly (and
eventually Firefox)
type=user
host=dynamic
hassip=yes
transport=ws,wss
dire...
2015 Jan 30
2
SSL traffic on RTP instance without an SSL session
Hi All
We've been reading this in the CLI a lot lately:
Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL
session
How can we find details about this particular RTP instance?
"rtp set debug" needs an IP which is precisely what I want to know (and I don't)!
Cheers
Ethy
2015 Aug 07
2
Asterisk 11.19.0 Now Available
...registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handsh...
2007 Oct 18
1
[simon@FreeBSD.org: cvs commit: src/crypto/openssl/ssl d1_both.c dtls1.h ssl.h ssl_err.c]
Hey,
RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail
beolow will not get a Security Advisory. This only affects
applications using DTLS, and I doubt there are many of those, but
users should still upgrade to get this fix, just in case.
See the OpenSSL advisory for some more details:
http://www.openssl.org/news/secadv_20071012.txt
If anybody were wondering, and hadn't checked the OpenSSL advisory:
older versions of FreeBSD are...
2007 Oct 18
1
[simon@FreeBSD.org: cvs commit: src/crypto/openssl/ssl d1_both.c dtls1.h ssl.h ssl_err.c]
Hey,
RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail
beolow will not get a Security Advisory. This only affects
applications using DTLS, and I doubt there are many of those, but
users should still upgrade to get this fix, just in case.
See the OpenSSL advisory for some more details:
http://www.openssl.org/news/secadv_20071012.txt
If anybody were wondering, and hadn't checked the OpenSSL advisory:
older versions of FreeBSD are...
2010 Oct 01
4
Patching openssl rpms
...6 (%prep)
The ssl.h.rej file has:
***************
*** 497,503 ****
/* SSL_OP_ALL: various bug workarounds that should be rather harmless.
* This used to be 0x000FFFFFL before 0.9.7. */
- #define SSL_OP_ALL 0x00000FFFL
/* DTLS options */
#define SSL_OP_NO_QUERY_MTU 0x00001000L
--- 497,503 ----
/* SSL_OP_ALL: various bug workarounds that should be rather harmless.
* This used to be 0x000FFFFFL before 0.9.7. */
+ #define SSL_OP_ALL
(0x000...
2015 Aug 07
0
Asterisk 13.5.0 Now Available
...hat
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
started when completing attended transfer (Reported by Joshua
Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handsh...