Table of Contents

Asterisk

Status: ✅ Full SIP/RTP flow automated in CI. The interop suite (tests/CalloraVoipSdk.InteropTests) runs against a real PJSIP Asterisk container (andrius/asterisk, via Testcontainers) on every relevant run:

  • Registration — 401→200 happy path, plus failure paths (wrong password / unknown user → prompt Failed, unreachable → Timeout).
  • Calls — outbound happy path (INVITE→200→ACK→live RTP→BYE), inbound (Asterisk→SDK), and failure paths (busy/486, decline/403, unknown/404, no-answer timeout, caller cancel).
  • Media — codec negotiation (PCMU/PCMA/G722 by preference), SRTP-SDES (RTP/SAVP + a=crypto, encrypted media), DTMF (RFC 4733) receive + negotiation.
  • In-call — hold/unhold (re-INVITE), blind & attended transfer (REFER/Replaces), session-timer negotiation (RFC 4028).
  • Early media (RFC 3960) — pre-answer receive-only media from a 180/183 SDP (plain and SRTP-SDES) plus DTMF in the early dialog.
  • Transport — UDP, TCP and TLS.
  • Two-leg bridged media — a separate suite bridges two SDK legs through the PBX and verifies bidirectional, byte-exact media (RTP counters both ways, local + remote RTCP quality, and byte-identical PCMU payload A→B).

The suite currently runs with all cases green and none skipped. Still, run your own acceptance test for anything you depend on before production.

Example pjsip.conf peer

[6001]
type = endpoint
context = from-internal
disallow = all
allow = ulaw,alaw,opus
auth = 6001-auth
aors = 6001

[6001-auth]
type = auth
auth_type = userpass
username = 6001
password = your-strong-password

[6001]
type = aor
max_contacts = 1

Connect from the SDK

var connect = await client.ConnectAsync(new SipAccount
{
    Username  = "6001",
    Password  = "your-strong-password",
    SipServer = "asterisk.lan"     // Asterisk host/IP
});

Expected configuration

  • Codecs — match allow= on the endpoint with your PreferredAudioCodecs; keep ulaw/alaw for a safe baseline.
  • DTMF — Asterisk defaults suit RFC 4733; keep dtmf_mode = rfc4733 on the endpoint.
  • SRTP — to require encryption set media_encryption = sdes on the endpoint and SrtpPolicy.Required on the SDK (see SRTP/SRTCP).
  • Reliable provisionals — the SDK only does 100rel when the peer requires it; set 100rel = required on the endpoint if you want PRACK.

Validate

Registration, internal dial between two endpoints, DTMF into an IVR/Read(), and SRTP if configured. Interop reports welcome: info@bechstein.digital.

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