1. RTP Real-Time Transport Protocol Speaker: Hsiao-Ting Wang Advisor: Quincy Wu Date: July 2 nd 2009

2. Outline Introduction RTP session RTP header  表頭欄位 Conclusion

3. Introduction UDP 缺點  封包的先後順序問題  如何知道封包有沒有遺失 RTP  RTP: A Transport Protocol for Real-Time Applications  It was first published in 1996 as RFC1889 which was made obsolete in 2003 by RFC 3550  想一想： RTP 比較適合 跑在哪個通訊協定上頭？ TCP or UDP? 連結層 實體層 網路層 (IP) 傳輸層 (TCP/UDP) 應用層 (HTTP 、 RTP)

4. Introduction (Cont.) RTCP(RTP Control Protocol)  RTCP provides out-of-band control information for an RTP flow RTCP 作用是提供 QoS (Quality of Service) 相 關的回饋 (feedback)  RTCP gathers statistics on a media connection and information such as bytes sent, packets sent, lost packets, jitter, feedback and round trip delay

5. RTP Session (RTP 會期 ) Consists of a number of applications communicating with RTP Identified by a network address and two ports  One port is used for media data  The other for RTCP (real time transfer control protocol) control data. 想一想：因此，打網路電話時，會佔去幾個 ports?

6. RTP Session (Cont.) Session participants can send, receive, or do both. Each media type is transmitted in a separate session, enabling participants to choose which media types they want to receive. For example, a user may just want the audio portion of a streaming music video. 想一想：那影像網路電話又會佔去幾個 ports?

7. RTP header 表頭大小為 12 bytes. IP headerUDP headerRTP headerRTP payload

8. 表頭欄位 Version (V)  2-bit. 目前的版本為 2. Padding (P)  1-bit. Payload 後面是否有 Padding. Extension (X)  1-bit. 有設的話，就會多加下面的延伸表頭.

9. 表頭欄位 (cont.) CSRC Count (CC)  4-bit. 0~15  在表頭裡包含 CSRC( Contributing source identifiers ) 的個數. Marker (M)  1-bit. 在 RFC1890 裡描述，在應用程式開始傳送封包時，第 一個封包應設定這個位元. Payload Type (PT)  7-bit. Defined by RFC 1990. Sequence Number  16-bit. 每次送出一個封包就加 1.  初始值為亂數值.  讓接收者用來偵測封包遺失與封包順序正確性.

10. 表頭欄位 (cont.) Timestamp  32-bit. Payload 裡的第一個 sample 產生時的時戳.  可用來做 jitter 的計算.  初使值由傳送端指定一亂數值. Synchronization Source (SSRC)  32-bit. The identifier is chosen randomly by the sender  Globally unique within an RTP session. Contributing Source (CSRC)  32-bit. 包含一個傳送端的 SSRC 值.  此欄位只有在來源從混音器 (mixer) 來時，才會使用.

11. RTP Header profile (RFC1900) ITU-T G.711

12. Wireshark

13. Graph Analysis

14. 名詞 Delay( 延遲 ) Jitter  Delay 的變化量

15. Delay 和 Jitter 示意圖 接收者根據收到的封包計算 delay 和 jitter 25ms 收到封包時間 播放內容時間 D 1 =75ms D 2 =50ms 50ms D 1 =100ms D 2 =100ms 目標 : 保持 jitter 越小越好 (0 in this case)

16. Conclusion RTP 作用是傳送 multimedia data

17. Reference RTP, http://en.wikipedia.org/wiki/Real- time_Transport_Protocol RTCP, http://en.wikipedia.org/wiki/Real- time_Transport_Control_Protocol