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Voice Codecs

Pick a codec and packetization (ptime) and I compute the bandwidth per call (with IP/UDP/RTP and layer-2 overhead), compared against other codecs by bitrate, MOS and delay. Multiply by the number of calls for the trunk.

The bitrate is the encoded voice rate. The MOS (1–5) estimates perceived quality.

How many ms of voice travel per packet. Fewer ms = more packets/s (pps) = more header overhead.

On top of the IP+UDP+RTP = 40 B headers you add the layer-2 frame (Ethernet, VLAN, MPLS…).

The trunk needs BW per call × number of calls.

redzilla.cl — codec 
 
  
 
 
BW per call
 
—
 
—
 
 
 
Trunk total
 
—
 
—
 
 
 
Quality (MOS)
 
—
 
—
 
 
  
 
 Per call —  
 
 Total —  
 
 Packet —  
 
  
 
Estimated quality (MOS)
 
 
 — scale 1–5 
 
 
 
—
 
  
 
Codec comparison
 

BW per call with the chosen packetization and overhead. The
          highlighted row is the selected codec. Tap a row to pick it.

 
 
  
 CodecBitrateSamplingFrame MOSDelayBW/call 
   
 
 
 
 How it is computed · payload, headers and pps 
 
 1. pps = 1000 / ptime. With
            20 ms of voice per packet that is 50 pps.

 
 2. payload (B) = bitrate_kbps × ptime_ms / 8. G.711 at
            64 kbps with 20 ms = 160 B.

 
 3. Per-packet headers:
IP 20 + UDP 8 + RTP 12 = 40 B plus layer 2
            (Ethernet 18 B, +VLAN 22 B,
MPLS 4 B).

 
 4. BW per call (bps) = (payload + 40 + L2) × 8 × pps.
            The total = BW per call × number of calls.

 
 5. MOS and
delay are reference values per codec; they do not
            depend on packetization but help you trade quality against
            bandwidth.

 
 
 
 
 
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