# Asterisk WebRTC + SipML5 no audio in Chrome

**URL:** <https://community.asterisk.org/t/asterisk-webrtc-sipml5-no-audio-in-chrome/70524>\
**Category:** Asterisk Support\
**Created:** [April 28, 2017, 11:50am UTC](https://community.asterisk.org/t/asterisk-webrtc-sipml5-no-audio-in-chrome/70524 "2017-04-28T11:50:46Z")\
**Posts on this page:** 2\
**Page:** 1

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**Author:** ![vinodv](https://avatars.discourse-cdn.com/v4/letter/v/97f17d/32.png) [@vinodv](https://community.asterisk.org/u/vinodv)\
**Post date:** [April 28, 2017, 11:50am UTC](https://community.asterisk.org/t/asterisk-webrtc-sipml5-no-audio-in-chrome/70524/1 "2017-04-28T11:50:46Z")

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Hi All,

I have a demo asterisk setup running in AWS Ec2 instance. We are using Asterisk 13.15.0 with WebRTC and SIpML5, but unfortunately, there’s no audio when calling from Chrome 58. Audio is fine when we try calling from Firefox and Zoiper. As per the recent update in the community I’ve updated the SipML5 to the latest, but still no luck. Below are my settings, can someone please help me to resolve the issue.

sip.conf

> [general]  
> externaddr= 52.247.186.5 ; Public ElasticIP  
> media\_address= 52.247.186.5 ; Public ElasticIP  
> localnet= 172.31.0.0/255.255.0.0 ; Internal network and mask  
> realm=52.247.186.5 ; Used for authentication  
> transport=udp,wss,ws,tls  
> websocket\_enabled = true  
> tlsenable=yes  
> tlsbindaddr=0.0.0.0  
> tlscertfile=/etc/ssl/certs/domain.pem  
> tlsprivatekey=/etc/ssl/private/domain.com.key  
> tlscafile=/etc/ssl/crt/domain.com-ca.crt  
> tlscipher=ALL  
> tlsclientmethod=tlsv1  
> nat=force\_rport,comedia  
> icesupport = yes  
> rtcp\_mux = yes

> [6000]  
> context=wrtc  
> host=dynamic  
> type=peer  
> encryption=yes  
> avpf=yes  
> icesupport=yes  
> directmedia=no  
> disallow=all  
> allow=ulaw  
> transport=ws,wss,udp,tls  
> force\_avp=yes  
> nat=force\_rport,comedia  
> secret=60000  
> dtlsenable=yes  
> dtlsverify=no  
> dtlscertfile=/etc/ssl/certs/domain.pem  
> dtlsprivatekey=/etc/ssl/private/domain.com.key  
> dtlscafile=/etc/ssl/crt/domain.com-ca.crt  
> dtlssetup=actpass  
> videosupport=no  
> rtcp\_mux = yes

> [6001]  
> context=wrtc  
> host=dynamic  
> type=peer  
> encryption=yes  
> avpf=yes  
> icesupport=yes  
> directmedia=no  
> disallow=all  
> allow=ulaw  
> transport=ws,wss,udp,tls  
> force\_avp=yes  
> nat=force\_rport,comedia  
> secret=6001  
> dtlsenable=yes  
> dtlsverify=no  
> dtlscertfile=/etc/ssl/certs/domain.com.pem  
> dtlsprivatekey=/etc/ssl/private/domain.com.key  
> dtlscafile=/etc/ssl/crt/domain.com-ca.crt  
> dtlssetup=actpass  
> videosupport=no  
> rtcp\_mux = yes

> ;extension to use on softphones  
> [6002]  
> host=dynamic  
> secret=6002  
> context=wrtc  
> type=peer  
> transport=ws,wss,udp,tls  
> directmedia=no  
> disallow=all  
> allow=all

rtp.conf

> [general]  
> rtpstart=10000  
> rtpend=20000  
> icesupport=yes  
> [stunaddr:19302=stun.l.google.com](http://stunaddr:19302=stun.l.google.com)

http.conf

> [general]  
> [servername=domain.com](http://servername=domain.com)  
> enabled=yes  
> bindaddr=0.0.0.0  
> tlsenable=yes  
> tlsbindaddr=0.0.0.0:8089  
> tlscertfile=/etc/ssl/certs/domain.com.pem  
> tlsprivatekey=/etc/ssl/private/domain.com.key

extensions.conf

> [wrtc]  
> exten =\> \_X.,1,Dial(SIP/${EXTEN})  
> exten =\> \_X.,n,Answer()  
> exten =\> \_X.,n,Hangup()

Sip debug can be found here [INVITE sip:6001@domain.com SIP/2.0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch - Pastebin.com](https://pastebin.com/ZNXEgsu5)

![](https://global.discourse-cdn.com/asterisk/original/2X/5/54ab97cb5aaad558eabdb1e2704e2bc63018f544.jpg)

Thank you

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<div class="post-metadata">

**Author:** ![tolja](https://avatars.discourse-cdn.com/v4/letter/t/eada6e/32.png) [@tolja](https://community.asterisk.org/u/tolja)\
**Post date:** [July 31, 2017, 11:14am UTC](https://community.asterisk.org/t/asterisk-webrtc-sipml5-no-audio-in-chrome/70524/2 "2017-07-31T11:14:34Z")

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Have you find a solution?, it is crazy, but i am trying the same thing, and i can’t get audio at all,  
cant get it even through zoiper to zoiper…