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Buffer at Wowza Server with RTMP streaming

I’m testing RTMP streaming with Wowza Server, from wowza sv1 over WAN to Wowza sv2.Obviously, i have to using repeater (origin & repeater-edge).The WAN,here WAN emulator,i can adjust jitter and delay, packet loss’s value for smooth RTMP streaming .

So, if i adjust Wowza server’s buffer (publishing & subscribing buffer, send & receive buffer), how that affect to jitter and packet loss for smooth RTMP streaming?

Welcome to Wowza forum.

Smooth playback is a function of stream bitrate / client bandwidth. If the quotient is < 1, you’re okay; If it is > 1, you will see buffering in the client. There is a BWCheck example that is will show you the actual server-to-client bandwidth, which changes from moment to moment as you will see. Optimal bitrate stream (lower than you would wish probably), or multi-bitrate streaming is the best way to mitigate.


What I mean is that the bitrate of a video has to less than the network capacity of a client, which is different for every client, and from moment to moment for each client. You will see “buffering” (stopping and starting, jerky playback) if the bitrate of the video is greater than client capacity. Changing buffer settings on server or client won’t really help.



Take look at /conf/VHost.xml /SocketConfiguration / Receive & SendBufferSize, which have inline comments and are discussed in the Wowza Tuning Guide:

What is the encoder? You said RTMP.

For MediaCaster sources (IP cameras, mpeg-ts encoder, etc), you can use Jitter Buffer and ModuleMediaCasterStreamMonitorAdvanced.

For RTMP push encoder, the best solution in this case is probably lower bitrate. You can monitor with something like this:


Hello Richard, thank for your support!

Do you mean subcribing buffer (at client)? What about publishing buffer (at wowza server) ? And when i adjust these buffer (at server & client), can the jitter and packet loss change ? I mean, with a set (jitter 1, packet loss 1) when these buffer are default (not adjust, config), streaming over WAN is ok (video & audio are nice), but when these buffer are changed,we can have a corresponding set (jitter 2, packet loss 2) ,obviously, streaming over WAN is also OK ?

Thank you!

So, with my poor network for streaming (bad jitter and packet loss value),i have to config buffer , time to packetizing and what parameters for solving, overcoming errors (because of my poor network) ? Will not changing buffer setting on server or client help to making streaming smooth with my poor network ?

Thank you for your support!