VoIP packet size

 Very good notes from learningnetwork.cisco.com


The best explanation is in chapter 5 of the Authorized Self-Study Guide for Cisco Voice over IP (CVoice) Second Edition by Kevin Wallace.  Basically, the formula is Bytes_per_sample = Sample_Size * CODEC_Bandwidth / 8 plus overhead.
So, if your sample size is 20 ms (.02 seconds)  and you are using the G.711 CODEC, then your basic voice information requires
.02 * 64000 / 8 = 160 bytes per sample.  To that, you must add the overhead, which would be 20 bytes for the IP header,
8 bytes for the UDP header, and 12 bytes for the RTP header (to make sure your packets are in the correct order at the receiving end).
So, each voice packet will require 200 bytes.  Then, you need to add your Layer 2 overhead (at least 18 bytes for Ethernet). 
So, each frame will require at least 218 bytes.  And you may also have trunk or tunneling overhead to consider.

Yes, voice packets are small, but you need a lot of them to carry voice.  That's why we use compression techniques.
cRTP compresses the IP/UDP/RTP header to either 2 or 4 bytes (4 if you implement check sum).  This significantly reduces the overhead.

In addition, you can use a CODEC that requires lower bandwidth.  The G.729 CODEC only requires 8000 bits per second, so if you used
G.729 and RTP header compression you would get
.02 * 8000 / 8 = 20 bytes of voice information plus 4 bytes for IP/UDP/RTP for an IP packet size of 24 bytes.  Then, you can add your
Layer 2 overhead, whatever it is.

Chapter 1 of the CCNP ONT Official Exam Certification Guide by Amir Ranjbar also contains a lot of good information.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
https://learningnetwork.cisco.com/thread/11162