Question

我试图从SIpJs打电话给Asterisk 12.我的同伴就在这里

[6002]
type=friend
secret=6002
host=dynamic
context=public
transport=ws
avpf=yes
icesupport=no
encryption = no

我的JsSip代码在这里

  var configuration = {
            'ws_servers': 'ws://192.168.0.102:8088/ws',
            'uri': 'sip:6002@192.168.0.102',
            'password': '6002'
        };
var options = {
            'eventHandlers': eventHandlers,
            'mediaConstraints': {'audio': true, 'video': false}
        };

        function call() {
            coolPhone.call('sip:6003@192.168.0.102', options);
        }

这是相应的注册，但当我打电话给＆＃34;打电话＆＃34;函数星号记录此错误

 Rejecting secure audio stream without encryption details: audio 46421 RTP/SAVPF 111 103 104 0 8 106 105 13 126

JSSIp错误在这里

调用失败，原因是：SDP不兼容

有人能帮助我吗？

Answer 1

首先，您需要为DTLS创建证书。然后从每个对等端启用DTLS。

使用以下命令创建证书。（将X.X.X.X替换为星号服务器IP）

mkdir /etc/asterisk/keys
cd ${ASTERISKSOURCE_PATH}/contrib/scripts/
./ast_tls_cert -C X.X.X.X -O "My Super Company" -d /etc/asterisk/keys

然后在同伴中添加以下按键：

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

将SipJs称为Asterisk 12

1 个答案: