我正在使用安装在云VPS上的RestComm Connect。
在进行基本测试呼叫以验证设置和连接时（使用默认的RestComm帐户Bob＆amp; Alice;在这种情况下Bob称为Alice），被叫方会报告以下错误：
Media path is lost due to connectivity issues; call has been hung up
然后在SIP＆＃39; 480暂时不可用＆＃39;之后呼叫失败。从被叫方发送给来电者。
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
RestComm的server.log摘录如下：
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>
这看起来像某种与ICE / STUN相关的NAT遍历问题。 我正在使用自定义STUN服务器并使用以下命令检查它是否有效：
vps404561:/$ stun 213.32.22.60:3478
STUN client version 0.96
Primary: Open
Return value is 0x000001
任何想法可能是这个问题的确切原因？ 我还在寻找一些提示/提示，以便进一步解决此问题。
谢谢，
多米尼克