我已嵌入JSSIP
http://tryit.jssip.net/打电话到我们的应用程序,它使用 Freeswitch
对于调用,除了调用之外的所有文字都会在 30 秒左右后断开连接,并且在浏览器 JS 控制台日志中我们看到以下内容,
在 Freeswitch
我看到来自 JSSIP
的重新邀请电话,目前Freeswitch
配置于 bypass_media=true
模式。
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:以上问题仅适用于 JSSIP 电话,它可以在 http://sipml5.org/ 上正常工作。网络电话。
最佳答案
对于手机来说,这是正常现象,这可能是非事件应用程序的操作系统限制。
对于 iOS 应用程序网络事件超时约为 30 秒。此应用程序之后网络请求将不会发送。
对于 Android 应用程序网络事件超时约为 30 秒到 3 分钟。
但请注意,关于 WebRTC Communications Consent :
Implementations MUST verify continuing consent at least every 30 seconds
关于javascript - JSSIP WebRTC 手机 30 秒后自动断开连接,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/30442224/