JSSIP WebRTC手机30秒后自动断开

Question

我已将JSSIP http://tryit.jssip.net/手机嵌入到我们的应用程序中，它使用Freeswitch进行呼叫，所有单词但在30秒左右后调用断开连接并且在浏览器JS控制台日志中我们是看到以下，

在Freeswitch方面，我看到来自JSSIP手机的重新邀请，目前Freeswitch模式配置bypass_media=true。

JS控制台登录浏览器：

JsSIP:InviteServerTransaction Timer L expired for transaction z9hG4bK9mjrH9cZ6FHtK +30s
jssip.js:21403 JsSIP:Transport received WebSocket text message:

BYE sip:50hn96ps@h1bf3jcld769.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WSS 10.20.20.212:7443;branch=z9hG4bKDSQUrNgDUKa5H
Max-Forwards: 70
From: "Satish" <sip:1003@10.20.20.212>;tag=6aQ2K8U19X09j
To: <sip:50hn96ps@h1bf3jcld769.invalid;transport=ws>;tag=5vuctmpuh3
Call-ID: 07a9b5e7-7d8e-1233-c2bf-2a1507b53463
CSeq: 75946179 BYE
User-Agent: FreeSWITCH-mod_sofia/1.4.18-3-1~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
Content-Length: 0


 +29s
jssip.js:21403 JsSIP:RTCSession receiveRequest() +12ms
jssip.js:21403 JsSIP:Transport sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 10.20.20.212:7443;branch=z9hG4bKDSQUrNgDUKa5H
To: <sip:50hn96ps@h1bf3jcld769.invalid;transport=ws>;tag=5vuctmpuh3
From: "Satish" <sip:1003@10.20.20.212>;tag=6aQ2K8U19X09j
Call-ID: 07a9b5e7-7d8e-1233-c2bf-2a1507b53463
CSeq: 75946179 BYE
Supported: outbound
Content-Length: 0


 +0ms
jssip.js:21403 JsSIP:RTCSession session ended +1ms
jssip.js:21403 JsSIP:RTCSession close() +0ms
jssip.js:21403 rtcninja:RTCPeerConnection close() +0ms
jssip.js:21403 JsSIP:RTCSession close() | closing local MediaStream +7ms
jssip.js:21403 rtcninja:Adapter closeMediaStream() | calling stop() on all the MediaStreamTrack +1ms
jssip.js:21403 JsSIP:Dialog dialog 07a9b5e7-7d8e-1233-c2bf-2a1507b534635vuctmpuh36aQ2K8U19X09j deleted +1ms
jssip.js:21403 JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bKDSQUrNgDUKa5H +2ms
jssip.js:21403 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: closed +0ms
jssip.js:21403 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: closed +1ms

更新：上述问题仅适用于JSSIP手机，适用于http://sipml5.org/网络电话。

Answer 1

对于手机这是正常的，这可能是非活动应用的操作系统限制。

对于iOS应用，网络活动超时约为30秒。此应用程序之后网络请求将不会发送。

对于Android应用，网络活动超时约为30秒至3分钟。

但请注意关于WebRTC Communications Consent：

  

实施必须至少每30个验证一次持续的同意   秒