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架構流程圖：分為傳送端、接收端、 GUI 三大部分

5-1 Transmitting RTP Media Streams：

5-2 Receiving RTP Media Streams：

5-3 GUI:

•  Media processing model：

傳送端：由 Capture 送出 data 經由 Process 送到網路。

接收端：由網路接收資料經由 Process 將 data 播放出來。

詳細過程會在以下說明，分為 Transmit 和 Receive。

5-1 Transmitting RTP Media Streams：

以下將會個別介紹各個 BLOCK 的功能及其操作流程。

5-1-1 Processor：

•  Produce an RTP-encoded DataSource.

•  Configuring the Processor.

•  Retrieving the Processor Output.

•  Controlling the Packet Delay.

5-1-2 Configuring the Processor：

•  Set RTP-specific formats for each track and specify the content descriptor you want.

•  Getting TrackControl for each track and calling setFormat to specify an RTP-specific format.

•  Example ： AudioFormat.GSM_RTP.

•  Output format ： setOutputContentDescriptor (method).

5-1-3 Retrieving the Processor Output：

•  Calling getDataOutput.

•  DataSource can be either a PushBufferDataSource or a PullBufferDataSource.

•  Connecting to the SessionManager using the createSendStream (method).

•  Multiple SourceStreams is sent out as a separate RTPstream, either in the same session or a different session.

5-1-4 Controlling the Packet Delay：

•  If the codec allows modification the interval.

•  Using the setPcketSize (method) of the packetizer codec's PacketSizeControl.

5-1-5 MediaLocator：

•  Calling Manager.createDataSink to construct an RTP DataSink.

•  Only transmit the first stream in the DataSource.

•  Provides the address and port of the RTP session.

5-1-6 Session Manager：

•  Creating send streams for the content and control the transmission.

•  To transmit multiple RTP streams in a session or to need to monitor session statistics.

5-1-7 Transmitting RTP Data using a DataSink：

•  Simplest way to transmit RTP data is to construct an RTP DataSink using Manager.creatDataSink (method).

•  You call start and stop on the DataSink.

•  Again, only the first stream in the DataSource is transmitted.

上圖即為 Transmit 的整個設計架構， Processor 處理來自 camera 的 data 之後 ，輸入對方 IP 將 data 送

到網路上等待接收端接收，其中 Audio 的 port 為 Video 的 port 加上 2 。Return to top

5-2 Receiving RTP Media Streams：

5-2-1 Session Manager：

5-2-2 MediaLocator：

接收主要可分為 2 個方式，一個是經由上面的路徑，也就是經由 MediaLocator 到 Data Source 到

Player 然後播放出檔案，另一個是經由下面的路徑，也就是經由 Session Manager 到 Data Source 到

Player 之後播放出檔案，在以下將會分析這 2 個路徑的不同。

在分析之前，先介紹一下何謂 Session Manager 和 MediaLocator ，及其功能：

5-2-1 Session Manager：

• keeps track of all of the participants in a session.

• used to coordinate an RTP session.

• maintain statistics on all of the RTP and RTCP packets sent and received in the session.

 

Session Manager 又包含了 2 個部份

• Session Statistics：

1. Statistics are tracked for the entire session on a per-stream basis.

2. GlobalReceptionStats：

   Maintains global reception statistics for the session.

3. GlobalTransmissionStats：

   Maintains cumulative transmission statistics for all local sender.

4. ReceptionStats：

   Maintains source reception statistics for an individual participant.

5. TransmissionStats：

   Maintains transmission statistics for an individual send stream.

• Session Streams：

1. eceiveStream：

   represents a stream that is being received from a remote participant.

2. SendStream：

   represents a stream of data coming from the Processor on input. Return

5-2-2 MediaLocator：

• An object that describes the media that a Player presents.

• Can be constructed even if the corresponding protocol handler is not installed on the system.

以下將介紹 2 種接收的方法：

第一種：經由 Session Manager 到 Data Source 到 Player 之後播放出檔案。

• Using this type you can deal with multiple RTP streams in the session.

• Construct a Player for each new receiveStream by retrieving the DataSource from the stream and passing it to Manager.createPlayer.

• When format or payload changes, the Player will not be able to handle the new format and JMF will throw an error if you attempt to present the new payload.

• ReceiveStreamListener needs to watch for RemotePayloadChangeEvent.

• When a RemotePayloadChangeEvent is detected, you need to：

• 1. Close the existing Player.

  2. Remove all listener for the removed Player.

  3. Create a new Player with the same RTP DataSource.

  4. Get the visual and control Components for the new Player.

  5. Add the necessary listeners to the new Player.

第二種：經由 MediaLocator 到 Data Source 到 Player 然後播放出檔案。

• Using this type you can only present the first RTP stream that is detected in the session.

• Construct a Player by calling Manager.createPlayer.

• When format or payload changes, Players automatically process payload changes. In most casts, this processing involves constructing a new Player to handle the new format.

以上 2 種方法不同之比較：

1. 使用第一種方法能在一個 session 中同時處理多個 RTP streams ，使用第二種則只能一次處理一個 RTP streams。

2. 當 payload 改變時，第一種傳輸方式的 Player 會無法應變，當你嘗試想要播放時就會出現錯誤。而第二種傳輸方式的 Player 當 payload 改變時它會自動的去建構一個新的 Player 使之能夠播放新的格式。

上圖即為 Receive 的整個設計架構，傳送端送出 data 後，接收端輸入自己的 IP 然後等待接收 data ，其中 Audio

的 port 為 Video 的 port 再加上 2 。Return Return to top

5-3 GUI:

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