VoIP.info Diagnostic

Asterisk PJSIP RTP address, secure or hold value is empty or wrong

Prove the current PJSIP leg, selected audio or video stream, bridge phase and observation time before changing NAT, direct-media, encryption or hold configuration.

Reviewed 2026-10-14Current Asterisk branch 23 CHANNEL documentation, Asterisk bridge and PJSIP troubleshooting guidance, and RFC 3264, RFC 3711 and RFC 6337 reviewed October 14, 2026; CHANNEL(rtp,...) reports media data for the exact current PJSIP channel and selected audio or video stream, while installed release, loaded modules, bridge technology, negotiated SDP, NAT translation, direct-media transition, keying, packet flow and audible result remain deployment specific; no PBX, module, endpoint, channel, bridge, SDP body, key, packet, media stream or call was inspected or testedAdvancedDocumentation based · not lab tested

Freeze the first mismatch

Record timestamp and timezone, installed Asterisk release, channel technology, name, unique ID and linked ID, call direction, bridge state, selector, explicit media type and returned value. Classify the symptom as empty src or dest, unexpected direct, wrong address or port, unexpected secure, unexpected hold, or disagreement with SDP or a packet capture.

Do not begin by changing NAT, direct-media or encryption settings. First preserve which channel and media interval produced the value.

Prove technology, leg and stream

Confirm CHANNEL(channeltype) is PJSIP and that dialplan is executing on the intended caller or callee leg. If video is requested, prove that the channel has the relevant video stream; if the media type is omitted, remember that Asterisk defaults to audio.

A blank value can be expected on another channel technology, the opposite leg, a stream that does not exist, or before the runtime state is available. Move the observation to the correct leg and interval before treating emptiness as a media failure.

Match the selector to its documented meaning

Compare src with the local RTP address and dest with the remote RTP address for that stream. Treat direct as its own direct-media remote address rather than a second spelling of dest. If direct media is not active, an empty direct value is not evidence that ordinary anchored RTP is broken.

Keep RTP addresses separate from CHANNEL(pjsip,local_addr) and remote_addr, which describe SIP INVITE sockets, and from URI, endpoint, Contact and DNS data.

Align bridge phase with SDP and packets

Determine whether the observation occurred before or after answer, re-INVITE, UPDATE, hold, resume, transfer, conference entry or a direct-media transition. Compare the offer-answer pair current at that time. RFC 3264 allows later exchanges to change media addresses and ports.

Capture at the interface that can actually see the expected path. With remote native bridging, endpoint-to-endpoint media may be outside Asterisk, so an Asterisk-host capture is not universal negative proof.

Keep encryption and hold conclusions narrow

For secure, compare the selected stream with negotiated media protection and observed SRTP evidence, while keeping signaling TLS, identity validation and the other bridge leg separate. For hold, compare the flag with SDP direction, current bridge behavior, RTP and the audible outcome.

A one flag proves neither end-to-end security nor successful hold audio. A zero flag does not identify which policy, negotiation, keying or topology stage caused the result.

Correct one demonstrated owner and repeat

Correct only the proven mismatch: observation leg, media type, dialplan timing, SDP address or port, NAT translation, endpoint policy, bridge feature, direct-media eligibility, encryption negotiation or hold workflow. Preserve the previous configuration and rollback.

Repeat the affected case and one known-good control. Acceptance requires the expected selector value on the exact stream plus matching SDP and packet evidence; separately verify two-way audible media. No PBX, channel, SDP body, key, packet or call was inspected or tested for this diagnostic.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

Examples require adaptation to your topology. No live PBX or hardware testing is claimed.