• 10-152018
  • Asterisk 16如何使用简捷Sip视频电话机召开视频电话

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    Asterisk 16 发布了全新的视频会议解决方案,具体可以参见《Asterisk 16 主要改进包括视频会议和sip电话性能》,那么今天我们要介绍的呢,就是怎样使用简捷的视频电话机来召开视频电话会议。


    首先呢,pjsip.conf配置

            pjsip的参数配置里面,主要是allow这个参数(Media Codecs to allow),要把支持的视频编码加进去。
            
            还有一个max_video_streams的参数,这个参数主要是设置视频的带宽,比如384kbps。

    然后重点就是ConfBridge这个应用了。

        ConfBridge(conference,[bridge_profile,[user_profile,[menu]]])

    在bridge_profile配置里,我们重点关注下面几个视频相关的参数:
    视频模式video_mode

    设置视频会议如何去分发视频给每个参会人员。 Note that participants wanting to view and be the source of a video feed MUST be sharing the same video codec. Also, using video in conjunction with with the jitterbuffer currently results in the audio being slightly out of sync with the video. This is a result of the jitterbuffer only working on the audio stream. It is recommended to disable the jitterbuffer when video is used.

    • none - No video sources are set by default in the conference. It is still possible for a user to be set as a video source via AMI or DTMF action at any time.
    • follow_talker - The video feed will follow whoever is talking and providing video.
    • last_marked - The last marked user to join the conference with video capabilities will be the single source of video distributed to all participants. If multiple marked users are capable of video, the last one to join is always the source, when that user leaves it goes to the one who joined before them.
    • first_marked - The first marked user to join the conference with video capabilities is the single source of video distribution among all participants. If that user leaves, the marked user to join after them becomes the source.
    • sfu - Selective Forwarding Unit - Sets multi-stream operation for a multi-party video conference.

    video_update_discard

    Sets the amount of time in milliseconds after sending a video update request that subsequent video updates should be discarded. This means that if we send a video update we will discard any other video update requests until after the configured amount of time has elapsed. This prevents flooding of video update requests from clients.

     

    remb_send_interval

    Sets the interval in milliseconds that a combined REMB frame will be sent to video sources. This is done by taking all REMB frames that have been received since the last REMB frame was sent, making a combined value, and sending it to the source. A REMB frame contains receiver estimated maximum bitrate information. By creating a combined REMB frame the sender of video can be influenced on the bitrate they choose, allowing better quality for all receivers.

     

    remb_behavior

    Sets how REMB reports are combined from multiple sources to form one. A REMB report consists of information about the receiver estimated maximum bitrate. As a source stream may be forwarded to multiple receivers the reports must be combined into a single one which is sent to the sender.

    • average - The average of all estimated maximum bitrates is taken and sent to the sender.
    • lowest - The lowest estimated maximum bitrate is forwarded to the sender.
    • highest - The highest estimated maximum bitrate is forwarded to the sender.


    最后我们关注一下哪些视频会议终端可以参与。如下的视频电话机是可以的。