Marker bit M : In audio streams , if silence suppression is enabled, the marker bit M SHOULD be one for the first packet of a talk spurt and zero for all other packets. Failure to do so can result in reduced audio quality at the receiving end. In video streams, the marker bit MUST be one for the last packet sent for each video frame , and zero for all other packets. Payload type PT : The payload type field identifies the format of the RTP payload , and determines its interpretation by the application.
Codecs that are not assigned to static payload types MUST be assigned to a payload type within the dynamic range, which is between 0x60 and 0x7f.
Codecs with payload type numbers in the dynamic range can use a different payload type number for send and receive. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. It receives voice and video streams from each participant, which may be encrypted using SRTP [RFC] , or extensions that provide participants with private media [I-D.
The goal is to provide a set of streams back to the participants which enable them to render the right media content. In a simple video configuration, for example, the goal will be that each participant sees and hears just the active speaker.
In that case, the goal of the switch is to receive the voice and video streams from each participant, determine the active speaker based on energy in the voice packets, possibly using the client-to-mixer audio level RTP header extension [RFC] , and select the corresponding video stream for transmission to participants; see Figure 1.
In order to properly support switching of video streams, the RTP switch typically needs some critical information about video frames in order to start and stop forwarding streams.
By providing meta-information about the RTP streams outside the encrypted media payload, an RTP switch can do codec-agnostic selective forwarding without decrypting the payload.
This document specifies the necessary meta-information in an RTP header extension. A subset of meta-information from the video stream is provided as an RTP header extension to allow an RTP switch to do generic selective forwarding of video streams encoded with potentially different video codecs.
The one-byte header format is used for examples in this memo. The two-byte header format is used when other two-byte header extensions are present in the same RTP packet, since mixing one-byte and two-byte extensions is not possible in the same RTP packet.
The separate specifications for Redundancy RTP Streams often include provisions for recovering any header extensions that were part of the original source packet.
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New post summary designs on greatest hits now, everywhere else eventually. Related 4. Hot Network Questions. Question feed. Home Overview Latest Changes. Overview Latest Changes. When two RTP streams are saved, you can select three options for way in which the audio streams in the capture are written as audio streams in the file: File synchronized - the saved audio starts at the beginning of the capture; both streams are prepended with silence to start them at the beginning of the capture.
The second stream is delayed to maintain the correct time relationship to the first one, so the two streams are synchronized. Stream synchronized - the saved audio starts at the beginning of the first stream. The first stream starts at the beginning of the audio file; the second stream is delayed to maintain correct the time relationship to the first one, so the two streams are synchronized.
Unsynchronized - both streams start at the beginning of the audio file, and the time relationship between the two streams is destroyed. Note: Waves were collected with audacity and combined with gimp. Other codec types It is possible to save in rtpdump format for any codec both audio and video and use e.
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