Table of Contents

FreeSWITCH

Status: 🧪 automated interop suite, run locally. FreeSWITCH is the second PBX behind the shared IPbxFixture abstraction: the two-leg scenario matrix that runs against Asterisk also runs against a real FreeSWITCH container (tests/CalloraVoipSdk.InteropTests, trait InteropFreeSwitch).

The suite is not part of the PR CI gate — it is a local-first check, so regressions are caught only when it is run explicitly:

dotnet test tests/CalloraVoipSdk.InteropTests -c Release --filter "Category=InteropFreeSwitch"

What is verified against FreeSWITCH

Each of these runs against a real FreeSWITCH container, with two SDK legs bridged through the PBX:

Scenario Assertion
Registration A directory user registers over UDP through the fixture adapter
Bridged call Both legs reach Connected
Media RTP flows in both directions, with RTP counters on each leg
Media content A marked PCMU payload arrives byte-exact A→B through the PBX
RTCP Local quality metrics and the remote SR/RR report are populated
SRTP-SDES An encrypted bridged call flows in both directions
Codec mismatch Legs with different codecs still flow, via transcoding
Hold / unhold Unhold resumes bidirectional media
Attended transfer The callee is bridged to a new target, with media flowing after the transfer
DTMF (RFC 4733) A digit traverses the bridge, caller → callee
Concurrent calls A short concurrent-call soak: all calls connect and flow media

What is not covered against FreeSWITCH

These are exercised against Asterisk only — do your own acceptance test before relying on them with FreeSWITCH:

  • Registration failure paths (wrong credentials, 423 Interval Too Brief)
  • Single-leg in/outbound calls and the codec-negotiation matrix (PCMU / PCMA / G722)
  • Blind transfer (REFER) as a separate scenario
  • Session timers (RFC 4028)
  • Early media (RFC 3960)
  • TCP and TLS transport

See the interop matrix for the full picture.

Directory user

A standard directory user (e.g. conf/directory/default/1001.xml) with a password works:

<user id="1001">
  <params>
    <param name="password" value="your-strong-password"/>
  </params>
  <variables>
    <variable name="user_context" value="default"/>
  </variables>
</user>

Connect from the SDK

var connect = await client.ConnectAsync(new SipAccount
{
    Username  = "1001",
    Password  = "your-strong-password",
    SipServer = "freeswitch.lan"    // the box running the internal SIP profile (5060)
});

Expected configuration

  • Profile — register against the internal profile unless you have a custom one.
  • Codecs — set the profile's inbound-codec-prefs / outbound-codec-prefs to overlap with your PreferredAudioCodecs (PCMU,PCMA is the safe baseline; add OPUS if enabled).
  • DTMF — RFC 4733 (rfc2833) is the default and matches the SDK.
  • SRTP — set the profile/dialplan to SDES and use SrtpPolicy.Required to enforce it.

Validate

Registration, a default-context dial to another extension, DTMF into an IVR, and SRTP if configured. Reports welcome: info@bechstein.digital.

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