G.711 Codec (PCMU and PCMA)
Short answer: G.711 is the ITU-T voice codec that encodes audio at 64 kbit/s using 8 kHz sampling, with two variants: µ-law (PCMU), used mainly in North America and Japan, and A-law (PCMA), used in most other regions. It is the baseline codec supported by virtually every SIP device.
Because G.711 applies almost no compression, it gives clear narrowband voice with minimal processing, at the cost of more bandwidth than compressed codecs.
How it works
Each audio sample is logarithmically companded to 8 bits. In RTP, PCMU uses static payload type 0 and PCMA payload type 8 (RFC 3551). With packet headers, a call typically needs around 80 to 90 kbit/s in each direction.
G.711 µ-law vs A-law
| Variant | RTP name | Static payload type | ZuteSIP |
|---|---|---|---|
| µ-law | PCMU | 0 | Supported |
| A-law | PCMA | 8 | Supported |
How it applies to ZuteSIP
ZuteSIP supports G.711 µ-law (PCMU) and A-law (PCMA). When configuring Zoiper, MicroSIP, Asterisk or another client for sip.zutesipdialer.com, enable at least one of these codecs so calls can negotiate.
Codecs: PCMU (G.711 u-law), PCMA (G.711 A-law)
Server: sip.zutesipdialer.com
Username: <SIP_USERNAME>Related terms
- SIP
- DTMF
- NAT and STUN
- Softphone
Frequently asked questions
Should I choose PCMU or PCMA?
Either works with ZuteSIP; enable both and let the call negotiate.
Why is there no audio with G.711 enabled?
One-way or no audio is usually a NAT or firewall issue rather than a codec problem.
Does ZuteSIP support Opus or G.729?
The listed codecs are G.711 PCMU and PCMA.
Related pages
- SIP
- DTMF
- NAT and STUN
- SIP troubleshooting
- set up a SIP dialer
- Asterisk integration
- CNAM (Caller Name Delivery)
- DID (Direct Inward Dialing) Number
Sources
- ITU-T: Recommendation G.711 (as of September 2026)
- IETF: RFC 3551, RTP Profile for Audio and Video Conferences (as of September 2026)