Re: [asterisk-users] webrtc no audio
are you sure you dont have this problem? https://issues.asterisk.org/jira/browse/ASTERISK-24146 i'm now fighting with https://issues.asterisk.org/jira/browse/ASTERISK-24602 Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a): I have it working now! *I had to install Asterisk 13 with PJSIP support.That's important, even if you won't use PJSIP.* Then I did this configuration, which is working fine under NAT: *sip.conf:* [6001] type=friend secret=REDACTED host=dynamic context=interno disallow=all ;allow=alaw,h263,h264,vp8 allow=g722 dtmf=auto videosupport=yes transport=ws,udp avpf=yes callerid=WebRTC 6001 encryption=yes qualify=yes directmedia=no nat=force_rport,comedia icesupport=yes 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 *rtp.conf:* icesupport=true stunaddr=stun.l.google.com:19302 http://stun.l.google.com:19302 *res_stun_monitor.conf:* stunaddr = stun.l.google.com:19302 http://stun.l.google.com:19302 ; Address of the STUN server to query.* * stunrefresh = 30 2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz: 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? You have to do a packet capture, look at the exchange in Wireshark, and see how the negotiation flows. It requires a basic understanding of ICE. it looks like we are facing this problem https://issues.asterisk.org/jira/browse/ASTERISK-24146 too if we use [] in sipml5 expert config To disable TURN/STUN to speedup ICE candidates gathering you can use an empty array. e.g. []. it works better -- --- 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 -- --- 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] webrtc no audio
I tested and it seems like I do have https://issues.asterisk.org/jira/browse/ASTERISK-24146 but in a different way. If I take more than 7s to answer the call, I don't get audio for a few seconds (about 3), after that it works okay. 2015-08-28 10:43 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz: are you sure you dont have this problem? https://issues.asterisk.org/jira/browse/ASTERISK-24146 i'm now fighting with https://issues.asterisk.org/jira/browse/ASTERISK-24602 Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a): I have it working now! *I had to install Asterisk 13 with PJSIP support.That's important, even if you won't use PJSIP.* Then I did this configuration, which is working fine under NAT: *sip.conf:* [6001] type=friend secret=REDACTED host=dynamic context=interno disallow=all ;allow=alaw,h263,h264,vp8 allow=g722 dtmf=auto videosupport=yes transport=ws,udp avpf=yes callerid=WebRTC 6001 encryption=yes qualify=yes directmedia=no nat=force_rport,comedia icesupport=yes 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 *rtp.conf:* icesupport=true stunaddr=stun.l.google.com:19302 *res_stun_monitor.conf:* stunaddr = stun.l.google.com:19302; Address of the STUN server to query. stunrefresh = 30 2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz: 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? You have to do a packet capture, look at the exchange in Wireshark, and see how the negotiation flows. It requires a basic understanding of ICE. it looks like we are facing this problem https://issues.asterisk.org/jira/browse/ASTERISK-24146 https://issues.asterisk.org/jira/browse/ASTERISK-24146 too if we use [] in sipml5 expert config To disable TURN/STUN to speedup ICE candidates gathering you can use an empty array. e.g. []. it works better -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 -- --- 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 -- _ -- 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] webrtc no audio
I have it working now! *I had to install Asterisk 13 with PJSIP support.That's important, even if you won't use PJSIP.* Then I did this configuration, which is working fine under NAT: *sip.conf:* [6001] type=friend secret=REDACTED host=dynamic context=interno disallow=all ;allow=alaw,h263,h264,vp8 allow=g722 dtmf=auto videosupport=yes transport=ws,udp avpf=yes callerid=WebRTC 6001 encryption=yes qualify=yes directmedia=no nat=force_rport,comedia icesupport=yes 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 *rtp.conf:* icesupport=true stunaddr=stun.l.google.com:19302 *res_stun_monitor.conf:* stunaddr = stun.l.google.com:19302; Address of the STUN server to query. stunrefresh = 30 2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz: 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? You have to do a packet capture, look at the exchange in Wireshark, and see how the negotiation flows. It requires a basic understanding of ICE. it looks like we are facing this problem https://issues.asterisk.org/jira/browse/ASTERISK-24146 too if we use [] in sipml5 expert config To disable TURN/STUN to speedup ICE candidates gathering you can use an empty array. e.g. []. it works better -- --- 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 -- _ -- 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] 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? You have to do a packet capture, look at the exchange in Wireshark, and see how the negotiation flows. It requires a basic understanding of ICE. it looks like we are facing this problem https://issues.asterisk.org/jira/browse/ASTERISK-24146 too if we use [] in sipml5 expert config To disable TURN/STUN to speedup ICE candidates gathering you can use an empty array. e.g. []. it works better -- --- 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] webrtc no audio
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? You have to do a packet capture, look at the exchange in Wireshark, and see how the negotiation flows. It requires a basic understanding of ICE. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] webrtc no audio
Marek Cervenka wrote: hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) BUT call from person2(chrome) to person 3(Jitsi sip client) - works! any tips howto find the problem? You would need to look at the ICE negotiation to see if it tried and failed. After that would be looking at the DTLS negotiation. Asterisk console output could provide some information. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] webrtc no audio
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?
-- Forwarded message --
From: Vinicius Fontes vinic...@aittelecom.com.br
Date: 2015-07-27 13:54 GMT-03:00
Subject: No audio on SIP over WebRTC
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.
In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.
Any tips on how to solve this? Here's my relevant files.
*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes
[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
*extensions.conf:*
[default]
exten = _6XXX,1,Dial(SIP/${EXTEN})
*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302
2015-08-10 12:35 GMT-03:00 Joshua Colp jc...@digium.com:
Marek Cervenka wrote:
hello,
i'm facing strange problem
asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
person1 to person3 are behind different NATs
audio devices double checked
call from person1(chrome) to person2(chrome) works
call from person1(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
BUT
call from person2(chrome) to person 3(Jitsi sip client) - works!
any tips howto find the problem?
You would need to look at the ICE negotiation to see if it tried and
failed. After that would be looking at the DTLS negotiation. Asterisk
console output could provide some information.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
--
_
-- 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
--
_
-- 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] webrtc no audio
hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) BUT call from person2(chrome) to person 3(Jitsi sip client) - works! any tips howto find the problem? -- --- 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
