G.711 is the uncompressed telephone codec at 64 kbit/s with the best voice quality and no licensing issues; G.729 compresses voice to 8 kbit/s, using far less bandwidth at a small cost in quality. In wholesale VoIP almost every carrier accepts G.711, and many also accept G.729, so the right choice depends on your bandwidth and on avoiding transcoding.
The two codecs compared
| Aspect | G.711 (PCMU / PCMA) | G.729 |
|---|---|---|
| Codec bitrate | 64 kbit/s | 8 kbit/s |
| Bandwidth per call, one direction (20 ms packets, IP, UDP and RTP headers) | About 80 kbit/s | About 24 kbit/s |
| Same, including Ethernet headers | About 87 kbit/s | About 31 kbit/s |
| Voice quality (typical MOS) | About 4.1 to 4.4 | About 3.9 |
| Fax and modem tones | Can pass, though T.38 is safer | Breaks them; use T.38 for fax |
| CPU load | Very low | Higher (compression) |
| Licensing | Free | Patents expired; now royalty free |
G.711 has two variants: PCMU (mu-law), used in North America and Japan, and PCMA (A-law), used in most other countries. Both are 64 kbit/s and most equipment handles either.
Bandwidth in practice
The headers matter more than people expect. At 20 ms packetization, a G.729 call carries 20 bytes of voice and 40 bytes of IP, UDP and RTP headers in every packet. That is why 8 kbit/s of voice becomes about 24 kbit/s on the wire. For capacity planning, multiply by the number of concurrent channels: 300 G.711 calls need around 26 Mbit/s each way with Ethernet overhead, while 300 G.729 calls need under 10 Mbit/s.
Quality
G.711 is the same quality as a traditional phone line. G.729 sounds slightly less natural but is fine for speech. The bigger quality risk is not the codec itself but changing codecs along the path. Each transcoding step adds delay and loses detail, and two G.729 compressions in a row sound noticeably worse. See MOS, jitter and packet loss for how quality is measured.
DTMF and fax
Compressed codecs distort the tones that IVRs and fax machines use. To keep DTMF working, carriers send key presses as separate events using RFC 2833 (now RFC 4733) rather than as audio. Agree the DTMF method during setup. For fax, use T.38 where both sides support it.
What carriers usually accept
- G.711 (both variants) is accepted almost everywhere.
- G.729 is widely accepted on wholesale routes, sometimes only G.729a (a lower complexity version that is compatible with G.729).
- Wideband codecs such as G.722 or Opus are rare on international wholesale routes, since the path to the mobile network usually ends in narrowband anyway.
Offer both G.711 and G.729 in your SIP setup and list them in order of preference. The SIP interconnect guide covers this exchange.
Which one to use
Use G.711 when bandwidth is not a constraint, which is the case for most data center links. Use G.729 when bandwidth is costly or limited, for example with remote call center sites. In both cases, match what your supplier uses to avoid transcoding. When you request a voice route, list your codecs and we source VoIP termination that accepts them.
Frequently asked questions
Is G.711 or Opus better?
Opus is a modern wideband codec with better quality per kilobit and strong tolerance of packet loss, and it is common in WebRTC and apps. On international wholesale routes it is rare, because the path to mobile networks usually ends in narrowband, so G.711 or G.729 is used there and Opus is converted at the edge.
What is the difference between G.711 and G.722?
G.722 is a wideband codec at 64 kbit/s that carries a wider range of audio frequencies than G.711, so voices sound clearer on calls that stay wideband end to end, such as between IP phones. Wholesale routes to the public network rarely support it, so calls to ordinary phones fall back to G.711 or G.729.
What is the difference between G.711 A-law and mu-law?
Both are 64 kbit/s versions of G.711 with the same quality, using different compression curves. Mu-law (PCMU) is used in North America and Japan, A-law (PCMA) in most other countries. Most equipment supports both, so offer both in your SIP setup to avoid a conversion between them.
Which VoIP codec uses the least bandwidth?
Among codecs common on wholesale routes, G.729 uses the least: 8 kbit/s of voice, or about 24 kbit/s per direction with IP, UDP and RTP headers at 20 ms packets. Lower bitrate codecs exist, but carriers rarely accept them. On VoIP links headers weigh a lot, so packetization settings also change the bandwidth used.
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