RTP
Earlier, we learned that the sender side of a VoIP application appends header elds to the audio
chunks before passing them to the transport layer. These header elds include sequence numbers
and timestamps. Since most multimedia networking applications can make use of sequence numbers
and timestamps, it is convenient to have a standardized packet structure that includes elds for
audio/video data, sequence number, and timestamp, as well as other potentially useful elds. RTP
is such a standard. RTP can be used for transporting common formats such as PCM, ACC, and
MP3 for sound and MPEG and H.263 for video. It can also be used for transporting proprietary
sound and video formats. Today, RTP enjoys widespread implementation in many products and
research prototypes. It is also complementary to other important real-time interactive protocols,
such as SIP.
RTP Basics
RTP typically runs on top of UDP. The sending side encapsulates a media chunk within an RTP
packet, then encapsulates the packet in a UDP segment, and then hands the segment to IP.
The receiving side extracts the RTP packet from the UDP segment, then extracts the media
chunk from the RTP packet, and then passes the chunk to the media player for decoding and
rendering. As an example, consider the use of RTP to transport voice. Suppose the voice source
is PCM-encoded (that is, sampled, quantized, and digitized) at 64 kbps. Further suppose that
the application collects the encoded data in 20-msec chunks, that is, 160 bytes in a chunk.
المادة المعروضة اعلاه هي مدخل الى المحاضرة المرفوعة بواسطة استاذ(ة) المادة . وقد تبدو لك غير متكاملة . حيث يضع استاذ المادة في بعض الاحيان فقط الجزء الاول من المحاضرة من اجل الاطلاع على ما ستقوم بتحميله لاحقا . في نظام التعليم الالكتروني نوفر هذه الخدمة لكي نبقيك على اطلاع حول محتوى الملف الذي ستقوم بتحميله .
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