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Call Codecs

Call codecs

Calls are controlled using the Session Initiation Protocol (SIP). Telephone calls are routed through SIP-to-PSTN gateways at our inbound and outbound providers. Call media is transported using the Real-time Transport Protocol (RTP). Within RTP, Aculab Cloud supports the G.711 A-law and G.711 mulaw audio codecs, and RFC2833 DTMF digits - all sampled at 8000Hz.

G.729 is available on request. Contact us if your need to use this codec with your provider.