Codecs & RTP

Behavior

Transcoding, packetisation and the media port range. Drag the codec list to change preference order globally. It is backed by codecs.conf · rtp.conf. It lives on the Media rail, under the Media & voice group: Codecs, RTP, recordings, prompts and conferencing.

Configuration

Codec preference

The order Asterisk offers codecs in an SDP. Drag to reorder — there is no list to type.

Global order (k_order) — a order control, default opus, g722, ulaw, alaw, g729.
*What it is:* The order codecs are offered in an SDP.
*Why it exists:* The far end picks the first one it also speaks, so order is preference.
*Choosing a value:* opus for quality, g722 for wideband on desk phones, ulaw as the universal fallback, g729 only where bandwidth is scarce and you have licences.
*Gotcha:* Putting a narrowband codec first means every call is narrowband, no matter what the phones support.
Allow transcoding (k_transcode) — a switch control, default true.
Opus bitrate (k_opusbr) — a slider control, default 24.
Preferred ptime (k_ptime) — a segmented control, default 20, choices 10, 20, 30, 40, 60.

RTP

Where media lands and how it survives a bad network.

RTP port range start (r_start) — a slider control, default 10000.
*What it is:* The lowest UDP port Asterisk will use for media.
*Why it exists:* Firewalls need to know the range to open.
*Choosing a value:* 10000 to 20000 is the usual convention.
*Gotcha:* Two calls need two ports each. A range smaller than four times your busy-hour concurrency will drop calls with no obvious error.
RTP port range end (r_end) — a slider control, default 20000.
RFC2833 payload (r_dtmf) — a stepper control, default 101.
strictrtp (r_strict) — a switch control, default true.
ICE support (r_ice) — a switch control, default false.
DTLS for WebRTC (r_dtls) — a switch control, default true.

Failure modes and security

Every control here maps to a real key in codecs.conf · rtp.conf; an unreachable configuration store is shown as unreachable, never backfilled with placeholder values. Putting a narrowband codec first means every call is narrowband, no matter what the phones support. Two calls need two ports each. A range smaller than four times your busy-hour concurrency will drop calls with no obvious error.

Verification

Exercise every control against its documented default and its full option range, confirm the write lands in codecs.conf · rtp.conf, and confirm an invalid combination is rejected before it reaches Asterisk. Confirm every default shown here matches what a fresh install actually ships, and that changing a value here is reflected the next time this screen loads.