[英]no audio with chrome using sip.js based webrtc app and asterisk 11.11.0. Working fine with firefox and Opera
我正在使用sip.js和星號開發webrtc。 我的webrtc應用程序在Firefox 31和Opera 22.0.1471.70上運行良好。 但是,當我將我的webrtc應用程序與chrome一起使用時(版本37.0.2062.58 beta-m(64位))。 一切似乎都很好,但是沒有音頻。 我還嘗試了最新穩定的Chrome瀏覽器的32版本。 問題保持不變。
Setup Details:-
1. Asterisk 32 bit - 11.11.0
2. Sip-0.6.1.js
3. Chrome (Version 37.0.2062.58 beta-m (64-bit)), Firefox 31.0 and opera 22.0.1471.70.
4. Cent O.S 6.5 (32 bit)
星號配置:
http.conf :-
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
sip.conf :-
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default
[1090] ; This will be WebRTC client
type=friend
username=1090 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=testsip ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
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
nat=no
disallow=all
allow=ulaw
extension.conf
[default]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061
webrtc應用程序代碼:-
call.js code:-
( function()
{
var session;
var endButton = document.getElementById('endCall');
endButton.addEventListener("click", function ()
{
session.bye();
alert ("Call Terminated");
}
, false
);
//Registration and websocket connectivity details for the useragent
var config = {
// Asterisk IP address
uri: '1090@192.168.56.129',
// Asterisk IP address,
// and replace the port with your Asterisk port from the http.conf file
wsServers: 'ws://192.168.56.129:8088/ws',
// Replace this with the username from your sip.conf file
authorizationUser: '1090',
// Replace this with the password from your sip.conf file
password: 'testsip',
// Enable sip traces on js console
traceSip: true,
stunServers: 'null',
};
//Creates the anonymous user agent so that you can make calls
var userAgent = new SIP.UA (config);
//Here you determine whether the call has video and audio
var options = {
media: {
constraints: {
audio: true,
video: false,
},
render: {
remote: {
audio: document.getElementById('remoteAudio')
},
local: {
audio: document.getElementById('localAudio')
}
}
}
};
function onAccepted ()
{
alert("Call Connected");
}
function onDisconnected ()
{
alert("Call Terminated");
}
//makes the call
session = userAgent.invite('1000', options);
session.on('accepted', onAccepted);
//session.on('disconnected', onDisconnected);
}
)();
SipCall.htm
<!DOCTYPE html>
<html>
<head>
<link rel="stylesheet" href="style.css">
</head>
<body>
<audio id="remoteAudio"></audio>
<audio id="localAudio" muted="muted"></audio>
<button id="endCall">End Call</button>
<script src="sip-0.6.1.js"></script>
<script src="call.js"></script>
</body>
</html>
請幫助我調試Chrome瀏覽器的無音頻問題。
任何幫助將不勝感激。
問候,
拉格文達·庫瑪(Raghuvendra Kumar)
您在星號服務器中啟用了vp8編解碼器,如果沒有,則嘗試從sip.conf中啟用vp8編解碼器。 如果是,則從瀏覽器將控制台日志復制到此處。
謝謝。!!!!
聲明:本站的技術帖子網頁,遵循CC BY-SA 4.0協議,如果您需要轉載,請注明本站網址或者原文地址。任何問題請咨詢:yoyou2525@163.com.