在 freeswitch 中捕获音频流
[英]Catch audio stream in freeswitch
问题描述
Given a sip call between two persons using freeswitch as my telephony engine ,how to catch audio stream of each person separately and process it before it's sent to the other end.
鉴于使用 freeswitch 作为我的电话引擎的两个人之间的 sip 呼叫，如何分别捕获每个人的音频流并在将其发送到另一端之前对其进行处理。 Thanks for your help in advance.提前感谢您的帮助。
1 个解决方案
解决方案1
1 2017-03-05 14:37:20
The only possible way i can think of is, Set up two conference.我能想到的唯一可能的方法是，设置两个会议。 Originate a call to A and connect to Conf A on answer.向 A 发起呼叫并在应答时连接到 Conf A。 call B and connect to Conf B. Now if A speaks you can record the call and convert to text - translate and convert to audio and play it to Conference B. Vice versa.呼叫 B 并连接到会议 B。现在，如果 A 发言，您可以记录通话并转换为文本 - 翻译并转换为音频并播放到会议 B。反之亦然。
ESL is a powerful module in Freeswitch where you can able to get all the events of freeswitch application and play with. ESL是 Freeswitch 中一个强大的模块，您可以在其中获取 freeswitch 应用程序的所有事件并进行播放。 In conference you get events when a member speaks, Joins, leaves, Mute and so on.在会议中，当成员发言、加入、离开、静音等时，您会收到事件。 Its an Idea but i've not tried it.它是一个想法，但我还没有尝试过。
Its like http://www.iamili.com/ that you gonna try :)它就像http://www.iamili.com/你要去尝试:)
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