作者： 默默

• begin_npt: PLAY返回消息里Range: npt=18.8-3600中的开始时间（秒）timestamp_len: RTP timestamp的最大表示个数，4个字节，所以是2^32即0x100000000
• reset_counter: RTP timestamp的归零次数
• current_timestamp: 当前收到的RTP的timestamp值
• start_timestamp: PLAY返回消息里的RTP-Info里的rtptime(指明此后第一个收到的RTP包的timestamp值)
• clock_rate: 时钟频率，即SDP a=rtpmap:96 H264/90000里的90000
• scale: PLAY里请求和返回中的Scale值（以返回为准）

[1] RFC2326 (Appendix B: Interaction with RTP):

For scaling (see Section 12.34), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 12.35) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.33) header provides
the first sequence number of the next segment.[2] 3GPP 26.234 (A.3.2.4 Timestamp handling after PAUSE/PLAY requests):

The description below intends to clarify how RTP timestamps are specified within the 3GPP PSS when a client sends a PLAY request following a PAUSE request. The RTP timestamp space must be continuous along time during a session and then reflect the actual time elapsed since the beginning of the session. A server must reflect the actual time interval elapsed between the last RTP packets sent before the reception of the PAUSE request and the first RTP packets sent after the reception of the PLAY request in the RTP timestamp. A client will need to compute the mapping between NPT time and RTP timestamp each time it receives a PLAY response for on-demand content. This means that a client must be able to cope with any gap in RTP timestamps after a PLAY request.

The PLAY request can include a Range header if the client wants to seek backward or forward in the media, or without a Range header if the client only wants to resume the paused session.

Example:
In this example Client C plays a media file from Server S. RTP timestamp rate in this example is 1000Hz for clarity.

C -> S: PLAY rtsp://example.com/mediastream RTSP/1.0
CSeq: 2
Session: 123456
Range: npt=1.125-

S -> C: RTSP/1.0 200 OK
CSeq: 2
Session: 123456
Range: npt=1.120-
RTP-Info: url=rtsp://example.com/mediastream;seq=1000;rtptime=5000

S -> C: RTP packet - seq = 1000 - rtptime = 5000 - corresponding media time (NPT time) = 1120ms
S -> C: RTP packet - seq = 1001 - rtptime = 5040 - corresponding media time (NPT time) = 1160ms
S -> C: RTP packet - seq = 1002 - rtptime = 5080 - corresponding media time (NPT time) = 1200ms
S -> C: RTP packet - seq = 1003 - rtptime = 5120 - corresponding media time (NPT time) = 1240ms

C -> S: PAUSE rtsp://example.com/mediastream RTSP/1.0
CSeq: 3
Session: 123456

S -> C: RTSP/1.0 200 OK
CSeq: 3
Session: 123456

[10 seconds elapsed]

C -> S: PLAY rtsp://example.com/mediastream RTSP/1.0
CSeq: 4
Session: 123456

S -> C: RTSP/1.0 200 OK
CSeq: 4
Session: 123456
Range: npt=1.280-
RTP-Info: url=rtsp://example.com/mediastream;seq=1004;rtptime=15160

S -> C: RTP packet - seq = 1004 - rtptime = 15160 - corresponding media time (NPT time) = 1280ms
S -> C: RTP packet - seq = 1005 - rtptime = 15200 - corresponding media time (NPT time) = 1320ms
S -> C: RTP packet - seq = 1006 - rtptime = 15240 - corresponding media time (NPT time) = 1360ms

C -> S: PAUSE rtsp://example.com/mediastream RTSP/1.0
CSeq: 5
Session: 123456

S -> C: RTSP/1.0 200 OK
CSeq: 5
Session: 123456

C -> S: PLAY rtsp://example.com/mediastream RTSP/1.0
CSeq: 6
Session: 123456
Range: npt=0.5-

[55 milliseconds elapsed during request processing]

S -> C: RTSP/1.0 200 OK
CSeq: 6
Session: 123456
Range: npt=0.480-
RTP-Info: url=rtsp://example.com/mediastream;seq=1007;rtptime=15295

S -> C: RTP packet - seq = 1007 - rtptime = 15295 - corresponding media time (NPT time) = 480ms
S -> C: RTP packet - seq = 1008 - rtptime = 15335 - corresponding media time (NPT time) = 520ms
S -> C: RTP packet - seq = 1009 - rtptime = 15375 - corresponding media time (NPT time) = 560ms

RTP包中的时间戳字段是说明数据包时间的同步信息，是数据能以正确的时间顺序恢复的关键。时间戳的值给出了分组中数据的第一个字节的采样时间。为了计算各个数据流的播放时间以及同步处理，仅有RTP包中的时间戳信息是不够的。
在整个播放过程中，包括这样几种时间:

一、时间戳映射关系

首先介绍PLAY请求的Response里的两个域：
（1）npt
npt顾名思义Normal PLay Time，正常播放时间，指出了流相对与播放开始时的绝对位置。播放开始时的之间定义为0.0s。这个特殊的常量now被定义为现场事件的当前时刻。"now"常数允许客户端请求接收实时反馈而不是存储或者延时的版本。因为对于这种情况而言,既没有绝对时间,也没有0时间,所以需要该参数。
（2）rtptime
rtptime是发送PLAY请求后将收到的第一个RTP包的时间戳值。
npt和rtptime的区别在于npt是影片开始的相对时间，而rtptime是会话开始的相对时间。因此在client端，需要对这两者进行map处理。
在client端计算播放时间戳的公式如下：

其中：
start_rtptime与start_npt分别是PLAY请求的Response中的rtptime和npt。
current_rtpTime是当前收到的RTP包头中的rtptime。
scale为PLAY请求的Response中的scale值。在正常播放的情况下为1，快速播放时大于1，当处于反向扫描模式时小于-1.

二、媒体间同步方法（不同设备的同步）

上面的处理仅仅实现了媒体内的同步，在实现媒体间同步时，还需要进行其他的处理工作。这就需要用到RTCP的SR(Sender Report)。在SR中包含一个<rtp，ntp>时间戳对，通过这个时间戳对可以将音频和视频准确的定位到一个绝对时间轴上。
<rtp，ntp>时间戳对的必要性在于不同流的RTP时间戳有不同的随机偏移量，因此无法直接进行同步。
<rtp，ntp>中的ntp是网络时间协议格式表示的绝对时间值。
<rtp，ntp>中的RTP与数据包中的RTP时间戳具有相同的单位和偏移量，但在大多数情况下此时间戳不等于任何临近的RTP包中的时间戳。
根据不同stream中的<rtp，ntp>时间戳对，可以将一个stream中的时间戳值映射为另一个stream的时间戳值，从而实现媒体间的同步。

参考链接

RTP时间映射及同步