Skip to main content

Media

The Bot runtime terminates the audio session. It validates the SDP offer, selects a compatible codec, allocates RTP and RTCP ports, and returns an SDP answer. RTP processing, jitter handling, playback, and provider audio conversion remain separate from SIP routing.

Codecs​

The runtime supports the following audio codecs:

CodecAudio rateRTP clockNotes
Opus48 kHz48 kHzVariable bitrate
PCMA8 kHz8 kHzG.711 A-law
PCMU8 kHz8 kHzG.711 μ-law
G.72216 kHz8 kHzWideband audio
L16SDP-definedSDP-definedLinear PCM

The default preference order is Opus, PCMA, PCMU, G.722, then L16. A call is rejected when the offer and the effective codec policy have no common codec.

Secure media​

Plain RTP and SDES-SRTP are supported. SRTP is deployment-controlled and is not automatically enabled for every Bot. When enabled with SDES, the runtime negotiates a compatible crypto suite, validates key material, and applies replay protection.

Supported suites are AEAD AES-128-GCM, AEAD AES-256-GCM, AES-128 Counter Mode with HMAC-SHA1-80, and AES-128 Counter Mode with HMAC-SHA1-32.

DTLS-SRTP keying is not implemented by the current RTP endpoint. Do not advertise DTLS-SRTP as a required media profile.