Webrtc 由于以下原因，重新通信调用失败："；“由于连接问题，媒体路径丢失”；错误

我正在使用安装在云VPS上的RestComm Connect

进行基本测试调用以验证设置和连接时（使用默认RestComm帐户Bob&Alice；在本例中，Bob称为Alice），被叫方报告以下错误：

Media path is lost due to connectivity issues; call has been hung up

然后在被呼叫方向呼叫方发送SIP“480暂时不可用”后，呼叫失败

浏览器控制台输出如下所示：

WebRTComm.js:4925 2017-10-10 17:06:31.370 WebRTCommCall:onRtcPeerConnectionIceChangeEvent(): Media path is lost due to connectivity issues; call has been hung up 

Stack trace: 
    at commonLog (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:4884:10)
    at console.error (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:4957:4)
    at WebRTCommCall.onRtcPeerConnectionIceChangeEvent (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:3871:12)
    at RTCPeerConnection.peerConnection.oniceconnectionstatechange (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:3047:9)
commonLog @ WebRTComm.js:4925
console.error @ WebRTComm.js:4957
WebRTCommCall.onRtcPeerConnectionIceChangeEvent @ WebRTComm.js:3871
peerConnection.oniceconnectionstatechange @ WebRTComm.js:3047
WebRTComm.js:4925 2017-10-10 17:06:31.399 WebRTCommCall:close(), received media stats 
    [Arguments.<anonymous> (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:2354:13)]
WebRTComm.js:4925 2017-10-10 17:06:31.400 WebRTCommCall:hangup() 
    [WebRTCommCall.hangup (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:2186:10)]
WebRTComm.js:4925 2017-10-10 17:06:31.401 PrivateJainSipCallConnector:close(): this.sipCallState=INVITED_INITIAL_STATE 
    [PrivateJainSipCallConnector.close (https://api-test.click2callme.pl/olympus/resources/js/mobicents-libs/WebRTComm.js:484:10)]
WebRTComm.js:4925 2017-10-10 17:06:31.402 SIP message sent: SIP/2.0 480 Temporarily Unavailable
Call-ID: 9b591496fb3bc5aac39075c5ea227d8b@213.32.22.60
CSeq: 1 INVITE
From: <sip:bob@77.255.251.128:57765>;tag=80839608_220982c6_57a5b08a_2ccbc3df
To: <sip:alice@178.43.83.112:63131>;tag=1507647967531
Max-Forwards: 70
Via: SIP/2.0/WSS 213.32.22.60:5083;branch=z9hG4bK2ccbc3df_57a5b08a_b9a9c4de-4268-4b16-a02b-4f8795cb4c2c;rport
Contact: <sip:alice@SpUEEE9qhnZn.invalid;transport=wss>
Content-Length: 0

17:04:22,641 INFO  [gov.nist.javax.sip.stack.SIPTransactionStack] (pool-AffinityJAIN-thread-47) <message
from="178.43.83.112:63131"
to="213.32.22.60:5083"
time="1507647862640"
isSender="false"
transactionId="z9hg4bk2ccbc3df_57a5b08a_b9a9c4de-4268-4b16-a02b-4f8795cb4c2c"
callId="9b591496fb3bc5aac39075c5ea227d8b@213.32.22.60"
firstLine="SIP/2.0 480 Temporarily Unavailable"
>
<![CDATA[SIP/2.0 480 Temporarily Unavailable
Call-ID: 9b591496fb3bc5aac39075c5ea227d8b@213.32.22.60
CSeq: 1 INVITE
From: <sip:bob@77.255.251.128:57765>;tag=80839608_220982c6_57a5b08a_2ccbc3df
To: <sip:alice@178.43.83.112:63131>;tag=1507647967531
Max-Forwards: 70
Via: SIP/2.0/WSS 213.32.22.60:5083;branch=z9hG4bK2ccbc3df_57a5b08a_b9a9c4de-4268-4b16-a02b-4f8795cb4c2c;rport
Contact: <sip:alice@178.43.83.112:63131;transport=wss>
Content-Length: 0

]]>
</message>

你知道这个问题的确切原因是什么吗？ 我也在寻找一些如何进一步解决这个问题的提示/技巧

谢谢

---

多米尼克

你好，多米。您是否可以共享呼叫者和被呼叫者的完整web控制台日志？如果在呼叫过程中对等方之间的媒体连接丢失，并且通常是由于网络连接不良导致的，对等方之间的击晕失败超过预定义的超时时间（我相信大约10秒），则会出现错误消息“由于连接问题导致媒体路径丢失”。另外，关于你的stun服务器，你是否确保客户端确实在使用它？嗨，多米。您是否可以共享呼叫者和被呼叫者的完整web控制台日志？如果在呼叫过程中对等方之间的媒体连接丢失，并且通常是由于网络连接不良导致的，对等方之间的击晕失败超过预定义的超时时间（我相信大约10秒），则会出现错误消息“由于连接问题导致媒体路径丢失”。另外，关于您的stun服务器，您是否确保客户机正在实际使用它？

vps404561:/$ stun 213.32.22.60:3478
STUN client version 0.96
Primary: Open
Return value is 0x000001