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On WSC 7.2.0, What Different H264 Levels For 3G And 4G Connection Can Be Used? (Doc ID 2357744.1)

Last updated on AUGUST 24, 2020

Applies to:

Oracle Communications WebRTC Session Controller - Version 7.2.0 to 7.2.0 [Release 7.2]
Information in this document applies to any platform.


This artical is to answer questions related to the following:

On 7.2.0 WSC, what different H264 levels for 3G and 4G connection can be used?
i.e. H.264 level 3.1 (1280x720@30 fps) for 4G and H.264 level 2.2 (720x576@12.5) or H.264 level 3 (720x576@25) for 3G from Application side.


Questions and Answers

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In this Document
Questions and Answers
 1) Is the SDK somehow prioritizing the audio stream?
 2) Is the SDK using a congestion control protocol?
 3) How does the SDK adapt the video quality based on network bandwidth? We can see that the video quality keeps changing depending on network conditions.
 4) How can we get networks statistics on RTP streams (eg latency, bit rate) for a P2P call?
 5) Can the application using the SDK modify the video frame rate?
 6) Can the application using the SDK modify the SDP adding for instance “b=AS:” so we can limit the bit rate?
 7) Is there a way to obtain networks statistics from the Media Engine on the RTP streams (eg latency, bit rate) for a established P2P audio/video call?
 8) If SDK received a video SDP offer from the conferencing server with “b=AS:”, would the SDK limit the video bitrate to max-bitrate as indicated in the b=AS: parameter?

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