We have successfully implemented the below configuration in a lab environment including simulated WAN. During configuration a switch from a variable bitrate stream (2-8Mbps with target 3.5Mbps) to constant 3.5Mbps caused issues which we resolved by upping the sortBuffer from 500 to 4000. This was a best guess rather than a scientific process.
Is there any documentation that lists all of these settings and their purpose? With the case described below can you suggest any improvements to the default config as per the setup instructions listed here
In particular I am interested in which settings are relevant in the rtp-live configuration when pulling a MPEG-TS and serving RTMP especially as latency and or packet drop increases.
(we have around 20 Wowza perpetual licenses and I can provide the final digits of the license key to support if required)
Hardware Encoder :Spinnaker 7100 HD publishing Mulitcast UDP MPEG-TS H.email@example.comMbps,1280x720 constant bit rate with each packet being 1316 in length (content)
Media Server: Wowza Media Server subscribing to multicast stream and serving RTMP and in some cases recording MP4 to disk
Client: Windows 7, Digital Signage Solution using Active X Flash Plugin to connect and playback rtmp stream
With udp streaming in from the encoder that you have, the packets can arrive out of order and the sort buffer buffers x milliseconds of packets and sorts them into the right order. Without this, it will just drop any late packets. It also aligns the video & audio as the audio packets are a lot smaller so an audio packet for a similar timecode to a video packet will arrive sooner.
You can also add a
jitter buffer which will help align everything and log packet loss in the log files.
This and the sort buffer both add to the latency of the stream so if that is an issue then you need to keep the settings as low as possible.