WebRTC No Audio
I’m facing strange problem:
Aasterisk13.5 + 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?
7 thoughts on - WebRTC No Audio
Marek Cervenka wrote:
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.
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 ——–
Vinicius Fontes wrote:
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.
Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
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
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:
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 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: