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iOS将RTP VoIP呼叫转移到后台

[英]iOS transfer RTP VoIP call to background

原文 2012-07-15 05:38:34 4 1 ios/ audio-streaming/ voip/ rtp

I need some help with a design decision. 我需要一些设计决策方面的帮助。 I am currently building an iOS app that requires VoIP capabilities and I have some questions. 我目前正在构建一个需要VoIP功能的iOS应用，我有一些疑问。

I understand that iOS can only maintain a TCP connection in the background. 我知道iOS只能在后台维护TCP连接。 If I have a RTP connection open during a call will that call be lost when the user enters the background? 如果我在通话过程中打开了RTP连接，当用户进入后台时，该通话会丢失吗？ If so how does Skype and others do this? 如果是这样，Skype和其他人如何做到这一点？
If I have to simply stream audio over TCP to keep a conversation going in the background, does anyone have a suggestion on how to keep the audio in sync in the event of the jitter buffer running empty during a call? 如果我仅需要通过TCP流音频以保持对话在后台进行，那么在通话过程中抖动缓冲区为空的情况下，是否有人提出了如何使音频保持同步的建议？

1 个解决方案

Answer to (1): 回答（1）：

iOS VoIP apps may keep a single TCP socket at background, usually your SIP socket, to allow incoming calls (INVITEs). iOS VoIP应用程序可能会在后台保留一个TCP套接字（通常是SIP套接字），以允许传入呼叫（INVITE）。

While your app has an active call, and goes to background, It has no restrictions , you can keep as many sockets as you want. 当您的应用程序正在进行通话并进入后台时， 它没有任何限制 ，您可以根据需要保留任意数量的套接字。

Here is a more detailed answer from another thread 这是另一个线程的更详细的答案

Regarding the SIP stack implementation, you may want to check open-source PJSip stack 关于SIP堆栈实现，您可能需要检查开源PJSip堆栈