[VoiceOps] High Quality, Reliable Voice via the Internet / SIPNOC

[Chris Boyd]

On Jun 9, 2014, at 9:00 PM, PE wrote:

> But isn't congestion caused by bytes and not number of packets? So, by that argument, larger packets will fill the queue faster than smaller and thus have a higher propensity to drop? And when it does, it is a bigger chunk of audio so it could actually reduce quality rather than improve it. 

Not always.....

If you look at the details of a lot of the hardware and software forwarding setups on network gear, you'll find that there's an outgoing buffer that packets go into if the port is already busy transmitting a frame.  Those buffers are typically sized such that they can hold a certain number of packets of MAX MTU for the device or interface, and it doesn't matter if it's 80 bytes of voice frame or an 8Kbyte jumbo frame going into the TX queue.

Of course QoS rules, congestion control mechanisms, WRED, and all sorts of other things can muck with the way a queue is serviced or filled, so YMMV, RTFM, etc.

--Chris