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vinicius at aittelecom...
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 Posted: Thu Aug 27, 2015 1:10 pm Post subject: [asterisk-users] webrtc no audio |
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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 <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
| Quote: | Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
| Quote: | Vinicius Fontes wrote:
| Quote: | 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
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cervajs at fpf.slu.cz
Guest
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| Posted: Fri Aug 28, 2015 8:43 am Post subject: [asterisk-users] webrtc no audio |
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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):
| Quote: | 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 <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
| Quote: | Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
| Quote: | Vinicius Fontes wrote:
| Quote: | 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
=======================================
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vinicius at aittelecom...
Guest
|
| Posted: Fri Aug 28, 2015 1:57 pm Post subject: [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 <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
| Quote: | 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):
| Quote: | 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 <cervajs@fpf.slu.cz (cervajs@fpf.slu.cz)>:
--
---------------------------------------
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
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