[英]Call quality metrics in sipML5
Does sipML provide any info about call quality? sipML 是否提供有关通话质量的任何信息? Something like dropped packets or packets arriving out of order?
诸如丢弃的数据包或乱序到达的数据包之类的东西?
I have looked at sipML API documentation, but did not find anything relevant.我查看了 sipML API 文档,但没有发现任何相关内容。 Also looked into the Developer Tools of Firefox/Chrome, but didn't find such metrics there.
还查看了 Firefox/Chrome 的开发者工具,但在那里没有找到这样的指标。
We would like to implement a call quality indicator similar to what other communication tools like Skype for Business or Teams have.我们希望实现与其他通信工具(如 Skype for Business 或 Teams)类似的通话质量指示器。
The sipML library uses two different protocols as the transport layer: sipML 库使用两种不同的协议作为传输层:
As SIP is agnostic to the transport layer, it doesn't bring any feature related to transport layers packets.由于 SIP 与传输层无关,因此它不会带来与传输层数据包相关的任何功能。 So there is no actual "out of the box" solution to detect quality of service.
因此,没有实际的“开箱即用”解决方案来检测服务质量。 The solution would be to have a server sending UDP packets back to the client in order to compute some metrics, but that requires some special implementation on both client and server side.
解决方案是让服务器将 UDP 数据包发送回客户端,以计算一些指标,但这需要在客户端和服务器端进行一些特殊的实现。
There doesn't seem to be an official API but if you can get a hold of the underlying RTCPeerConnection object ( here? ) you can use the full getStats() API似乎没有官方的 API 但如果你能掌握底层的 RTCPeerConnection object ( 这里? )你可以使用完整的getStats() ZDB974238714CA8DE634A7CE1D083A14
声明:本站的技术帖子网页,遵循CC BY-SA 4.0协议,如果您需要转载,请注明本站网址或者原文地址。任何问题请咨询:yoyou2525@163.com.