In order to improve publishing performance under poor network conditions, we have added support for RTC publishing in addition to the traditional RTMP protocol. This document compares the performance of publishing under different network conditions using the two protocols.
Note
Performance Under Normal and Poor Network Conditions
Test method
We simulated different network conditions at the publishing end and observed the playback effects (the streams were played over CDNs, and network conditions at the playback end were normal).
Parameter configuration
To prevent the use of different sources from affecting the results, we used V2TXLivePusher to publish the same video over RTMP and RTC.
Video parameters:
Parameter | Value |
Resolution | 720 x 1280 |
Bitrate | 1800 Kbps |
Frame rate | 15 |
Comparison of performance under different network conditions
Frame rate
Note
Stutter rate
Note
Appendix: Network Parameters
Parameter | Description |
Frame rate | Frames rendered per second |
Packet loss | A packet loss rate of 50% means that for every 10 data packets sent, five fail to arrive at their destination. |
Latency | A latency of 200 ms indicates that data packets are delivered by the network only 200 ms after they are sent by the SDK. |
Transfer speed cap | A transfer speed cap of 800 Kbps means that 800 KB of data can be sent per second at most. |
Stutter rate | Stutter occurs if the interval between the rendering of two consecutive frames exceeds 200 ms. Stutter rate is the total stuttering time divided by the total playback time. |
Yes
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