我已经将JSSIP http://tryit.jssip.net/电话嵌入到我们的应用程序中，它使用Freeswitch进行呼叫，除了呼叫在30秒左右后断开连接之外的所有内容，在浏览器JS控制台日志中我们看到以下内容，
在Freeswitch方面，我看到来自JSSIP电话的重新邀请，目前Freeswitch配置在bypass_media=true模式。
JS控制台日志在浏览器:
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:以上问题仅适用于JSSIP电话，它可以与http://sipml5.org/Webphone一起工作。
对于手机来说这是正常的，这可能是操作系统对非活动应用的限制。
对于iOS应用程序，网络活动超时时间约为30秒。在此应用程序之后，网络请求将不再发送。
Android应用的网络活动超时时间约为30秒至3分钟。
但请注意关于WebRTC通信同意:
实现必须至少每30次验证持续同意秒