Re: [asterisk-users] Asterisk 13 WebRTC Status report
Den 2015-09-15 kl. 16:52, skrev asandoval...@gmail.com: Hello Marek! I’ve been running on an issue with my Asterisk 12 configuration for using WebRTC on a LAN environment for about a month! I really need some help … My calls from the browser are done fine. I get ringing, they can be answered and never drop. The thing is that there is no audio on any side! But I don’t get any error or warning from JavaScript nor the Asterisk CLI. I’m using Asterisk 12 + jsSIP. If you could help me solving this I would be eternally greatful It’s for my grade project … These are my files: sip.conf: http://pastebin.com/kWwXpi4V http.conf: http://pastebin.com/ZwJWiiwf SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb SIP debugging for extension call (Hello-World recording): http://pastebin.com/0PxjLwBb I followed these tutorials. If you have any other useful resource, I’d be glad if you could share it: http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11 http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html Furthermore, if I want to have a local Asterisk configuration, which should be the IP address for the http.conf + DTLS certificates?? I tried with localhost but RTP packets redirect to my eth IP. Thanks in advance!! In asterisk you have "rtp set debug on" to see if you get rtp packets. On your client you can start wireshark and look if RTP packets flow in both directions. If you have RTP traffic, maybe you didn't attach the incoming media to an audio/video tag in your html. For example: html: In the event-handler for 'addstream' for the call, you have to attach the stream to #remoteView. /Johan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 WebRTC Status report
Hello Marek! I’ve been running on an issue with my Asterisk 12 configuration for using WebRTC on a LAN environment for about a month! I really need some help … My calls from the browser are done fine. I get ringing, they can be answered and never drop. The thing is that there is no audio on any side! But I don’t get any error or warning from JavaScript nor the Asterisk CLI. I’m using Asterisk 12 + jsSIP. If you could help me solving this I would be eternally greatful It’s for my grade project … These are my files: sip.conf: http://pastebin.com/kWwXpi4V http.conf: http://pastebin.com/ZwJWiiwf SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb SIP debugging for extension call (Hello-World recording): http://pastebin.com/0PxjLwBb I followed these tutorials. If you have any other useful resource, I’d be glad if you could share it: http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11 http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html Furthermore, if I want to have a local Asterisk configuration, which should be the IP address for the http.conf + DTLS certificates?? I tried with localhost but RTP packets redirect to my eth IP. Thanks in advance!! De: Marek Červenka Enviado el: martes, 15 de septiembre de 2015 06:37 a. m. Para: asterisk-users@lists.digium.com hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from https://issues.asterisk.org/jira/browse/ASTERISK-24106 chan_sip is not usable for webrtc because of https://issues.asterisk.org/jira/browse/ASTERISK-24602 another problem arise with RTP/SAVPF negotiation this can be solved with patch for Asterisk from https://issues.asterisk.org/jira/browse/ASTERISK-24602 and for pjsip http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html i hope this info helps what is your experience with WebRTC? See you at WebRTC Expo Paris :) p.s. many thanks to my colleague martin tomec for debugging support p.s.2 relevant part of pjsip.conf [global] [transport-wss] type=transport protocol=wss;udp,tcp,tls,ws,wss bind=0.0.0.0 ;===ENDPOINT TEMPLATES [endpoint-basic](!) type=endpoint transport=transport-wss context=route_phones disallow=all allow=alaw allow=ulaw force_avp=yes use_avpf=yes; Determines whether res_pjsip will use and enforce usage of media_encryption=dtls; Determines whether res_pjsip will use and enforce dtls_verify=no ; Verify that the provided peer certificate is valid (default: dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey dtls_cert_file=/etc/pki/tls/certs/pbx.crt dtls_private_key=/etc/pki/tls/private/pbx.key dtls_setup=actpass ice_support=yes ;This is specific to clients that support NAT traversal media_use_received_transport=yes [auth-userpass](!) type=auth auth_type=userpass [aor-single-reg](!) type=aor remove_existing=yes max_contacts=1 ;===DEVICES [webrtc1](endpoint-basic) auth=webrtc1 aors=webrtc1 [webrtc1](auth-userpass) password=secret username=webrtc1 [webrtc1](aor-single-reg) relevant part of http.conf [general] enabled=yes bindaddr=0.0.0.0 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/pki/tls/certs/pbx.crt tlsprivatekey=/etc/pki/tls/private/pbx.key -- --- Marek Cervenka ===-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13 WebRTC Status report
hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from https://issues.asterisk.org/jira/browse/ASTERISK-24106 chan_sip is not usable for webrtc because of https://issues.asterisk.org/jira/browse/ASTERISK-24602 another problem arise with RTP/SAVPF negotiation this can be solved with patch for Asterisk from https://issues.asterisk.org/jira/browse/ASTERISK-24602 and for pjsip http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html i hope this info helps what is your experience with WebRTC? See you at WebRTC Expo Paris :) p.s. many thanks to my colleague martin tomec for debugging support p.s.2 relevant part of pjsip.conf [global] [transport-wss] type=transport protocol=wss;udp,tcp,tls,ws,wss bind=0.0.0.0 ;===ENDPOINT TEMPLATES [endpoint-basic](!) type=endpoint transport=transport-wss context=route_phones disallow=all allow=alaw allow=ulaw force_avp=yes use_avpf=yes; Determines whether res_pjsip will use and enforce usage of media_encryption=dtls; Determines whether res_pjsip will use and enforce dtls_verify=no ; Verify that the provided peer certificate is valid (default: dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey dtls_cert_file=/etc/pki/tls/certs/pbx.crt dtls_private_key=/etc/pki/tls/private/pbx.key dtls_setup=actpass ice_support=yes ;This is specific to clients that support NAT traversal media_use_received_transport=yes [auth-userpass](!) type=auth auth_type=userpass [aor-single-reg](!) type=aor remove_existing=yes max_contacts=1 ;===DEVICES [webrtc1](endpoint-basic) auth=webrtc1 aors=webrtc1 [webrtc1](auth-userpass) password=secret username=webrtc1 [webrtc1](aor-single-reg) relevant part of http.conf [general] enabled=yes bindaddr=0.0.0.0 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/pki/tls/certs/pbx.crt tlsprivatekey=/etc/pki/tls/private/pbx.key -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 WebRTC Status report
Dne 15.9.2015 v 13:37 Marek Červenka napsal(a): hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from https://issues.asterisk.org/jira/browse/ASTERISK-24106 chan_sip is not usable for webrtc because of https://issues.asterisk.org/jira/browse/ASTERISK-24602 this is the blocking issue https://issues.asterisk.org/jira/browse/ASTERISK-24146 -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
