SIP with AudioSocket

Asterisk 18 and later can stream a call's audio over TCP with AudioSocket(). Spiiksi listens for it, plays the line as one side of a call, and plays the other side's interpretation back into the call. No Twilio needed.

Inbound calls: an entry id#

1. In Spiiksi, open API & integrations → Connectors → SIP / PBX. Choose your agents' language and the callers' language, and Connect.
2. Copy the entry id (shown once; make a new one if it's lost).
3. Add a dialplan context, and route calls to it from an extension, an IVR choice or a queue overflow:

[spiiksi-inbound]
exten => 500,1,Answer()
 same => n,Set(SPIIKSI_ENTRY=00000000-0000-0000-0000-000000000000)
 same => n,AudioSocket(${SPIIKSI_ENTRY},audio.spiiksi.fi:9092)
 same => n,Hangup()

Each call to it starts a new Spiiksi call with the caller as the customer. Agents join from Calls.

Callers who speak different languages: let your IVR ask, then start a call with that customer_language through the API and pass the side's line id (below); or route through Twilio, whose menu asks the caller.

A side of a call you started#

Start a call with the API, then ask for a side's own line id:

curl https://api.spiiksi.fi/v1/calls/$CALL_ID/phone-line \
  -H "Authorization: Bearer $SPIIKSI_KEY" -H "Content-Type: application/json" \
  -d '{"side": "customer"}'

{"audiosocket_id": "3f2a…", "host": "audio.spiiksi.fi", "port": 9092}

Pass it to AudioSocket(3f2a…,audio.spiiksi.fi:9092), for example from an AGI script that started the call. A new id replaces the side's old one.

The connection#

• TCP to audio.spiiksi.fi:9092; allow it outbound from the PBX.
• Asterisk's AudioSocket protocol: a 16-byte UUID first, then 16-bit 8 kHz mono audio frames, both ways, and a hangup at the end.
• Audio isn't encrypted on this link. Run it over a VPN, or use Twilio's TLS/SRTP SIP if your calls need encryption on the way.

FreeSWITCH and others can use an AudioSocket-compatible module, or reach Spiiksi through Twilio.