Calculating VoIP Throughput Requirements

Member rakem recently mentioned in the forums an issue he experienced with a QoS policy for VoIP traffic over a low-bandwidth (512 Kbps) link. The original QoS policy had allocated 15% of the available throughput for VoIP traffic. Generally speaking, this is not an unacceptable approach; however, such slow links require diligent consideration of absolute (versus relative) traffic requirements.

A QoS policy map statement which allocates 15% of the available throughput on a 512 Kbps interface is only reserving roughly 76 Kbps for the given traffic class. Such a policy might be configured like this:

class-map match-all VoIP
 match  dscp ef 
!
policy-map Uplink
 class VoIP
  priority percent 15
!
interface FastEthernet0/0
 bandwidth 512
 service-policy output Uplink

We can use the command show service-policy interface to verify that the policy is applied as expected:

Router# show policy-map interface f0/0
 FastEthernet0/0

Service-policy output: Uplink

Class-map: VoIP (match-all)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      Match:  dscp ef (46)
      Queueing
        Strict Priority
        Output Queue: Conversation 136 
        Bandwidth 15 (%)
        Bandwidth 76 (kbps) Burst 1900 (Bytes)
        (pkts matched/bytes matched) 0/0
        (total drops/bytes drops) 0/0

Class-map: class-default (match-any)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      Match: any 

In rakem's case, the VOIP codec being used was G.711, which requires 64 Kbps of throughput per call. However, this number accounts only for voice payload. To calculate the overall throughput required per call, we have to factor in payload sample size (packetization rate) and lower-layer packet headers (RTP, UDP, IP, and layer two). A sample calculation can be found on the VoIP Basics cheat sheet. We can calculate this overhead manually or use a VoIP bandwidth calculator like the one at bandcalc.com:

Without factoring in the L2 overhead (which will vary depending on the medium used), we can calculate that a single VoIP call using the G.711 codec and a 20 msec sample size requires at least 80 Kbps of throughput (each way), more than the 76 Kbps allocated by the 15% QoS policy. Packets from the voice stream will be dropped when there is contention for the available throughput.

Using the VoIP bandwidth calculator and specifying Ethernet (non-802.1Q) as the layer two medium, we see that a single call requires at least 95.2 Kbps of throughput, which we'll round up to 96 Kbps. Now we can redefine the earlier QoS policy with a more appropriate measure:

class-map match-all VoIP
 match  dscp ef 
!
policy-map Uplink
 class VoIP
  priority 96
!
interface FastEthernet0/0
 bandwidth 512
 service-policy output Uplink

Of course, we're still limited to a single call of guaranteed quality. A more desirable solution, as rakem saw, would be to employ a more efficient voice codec. G.729a, for example, requires only 8 Kbps of throughput for its voice payload. This results in a total unidirectional utilization of ~40 Kbps per call (assuming Ethernet is used at L2); less than half of that required for a single G.711 call.