search for: dtlsprivatekey

Displaying 9 results from an estimated 9 matches for "dtlsprivatekey".

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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...