Displaying 9 results from an estimated 9 matches for "dtlsprivatekey".
Did you mean:
tlsprivatekey
2016 Oct 05
2
Ast 13.10 to 13.11 stop working webrtc
...guration
of the module using TLS.
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in sip.conf, for example:
dtlscipher=AES128-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, e.g., sip.conf dtlsprivatekey. For example:
openssl dhparam -out ./dh.pem 2048
- Because clients expect the server to prefer PFS, and because OpenSSL sorts
its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
Consider re-ordering your cipher suites in the respective configuration
file. For example:
dt...
2015 Oct 28
2
Receiving Messages and Extensions Config for WebRTC
...t=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 is
;dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when
setting up DTLS
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=fee50
;encryption=yes ; Tell Asterisk to use...
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
...contactpermit
contactdeny
fullname 660 win8
hasvoicemail
subscribemwi
dtlsenable yes
dtlsverify no
dtlscertfile /etc/asterisk/keys/asterisk.pem
dtlsprivatekey /etc/asterisk/keys/asterisk.pem
dtlssetup actpass
sippasswd md5pwd
rpid
domain testers.com
sippasswd2
and my sip.conf:
[general]
bindport = 5070
bindaddr = PU.BL.IC.IP
udpbindaddr =...
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?
>
2014 Aug 11
1
Letting rtp profiles be handled by rtpengine instead of Asterisk
....BL.IC.IP
contactpermit: NULL
contactdeny: NULL
fullname: NULL
cid_number: NULL
callingpres: NULL
mohinterpret: NULL
mohsuggest: NULL
hasvoicemail: NULL
subscribemwi: NULL
vmexten: NULL
rtpkeepalive: NULL
directrtpsetup: yes
dtlsenable: yes
dtlsverify: no
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
dtlssetup: actpass
dtlscertfile: /etc/asterisk/keys/asterisk.pem
dtlscafile: /etc/asterisk/keys/ca.crt
sippasswd: md5ofmypwd
rpid: NULL
domain: testers.com
sippasswd2: NULL
This is how all other clients are currently defined:...
2014 Aug 06
1
From and To headers contain same account in INVITEs
...: 1.1.1.1
contactpermit: NULL
contactdeny: NULL
fullname: NULL
cid_number: NULL
callingpres: NULL
mohinterpret: NULL
mohsuggest: NULL
hasvoicemail: NULL
subscribemwi: NULL
vmexten: NULL
rtpkeepalive: NULL
directrtpsetup: yes
dtlsenable: yes
dtlsverify: no
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
dtlssetup: actpass
dtlscertfile: /etc/asterisk/keys/asterisk.pem
dtlscafile: /etc/asterisk/keys/ca.crt
sippasswd: a84a4ddcda13d13c9573d5294047b6a2
rpid: NULL
domain: testers.com
sippasswd2: NULL
id: 8
name...
2015 May 21
1
asterisk 13 webrtc
...an't provide secure audio requested in
SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
canreinvite=no
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=yes
transport=wss,ws
dtlsrekey=60
dtlsverify=no
dtlscertfile=/etc/pki/tls/certs/rapidssl.crt
dtlsprivatekey=/etc/pki/tls/private/rapidssl.key
dtlssetup=actpass
sip dump
<--- SIP read from WS:2.2.2.2:8558 --->
INVITE sip:887 at ipbx SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport
From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;...
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001 at device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no
With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk...
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
...NULL
regexten: NULL
regserver:
fromdomain: testers.com
videosupport: no
contactpermit: NULL
contactdeny: NULL
fullname: 660 win8
subscribemwi: NULL
dtlsenable: yes
dtlsverify: no
dtlscertfile: /etc/asterisk/keys/asterisk.pem
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
dtlssetup: actpass
sippasswd: a84a4ddcda13d13c9573d5294047b6a2
rpid: NULL
domain: testers.com
sippasswd2: 5c4671ae1043e6116118fed39bee091a
callbackextension: NULL
insecure: NULL
cheers,
Olli
-------------- ne...