How to use Telnyx as a Termination Carrier for Plivo

Users with accounts in both Telnyx and Plivo portals can send calls from Plivo to Telnyx via SIP and have Telnyx terminate those calls on the PSTN.

Introduction

The following instructions will explain how to configure both sides so that calls from Plivo are authenticated using your Telnyx SIP Connection.

In this guide, we use a Plivo XML application as an example, but the token configuration on Plivo side will follow the same analogy for any other Plivo API based application (SDK, node, python, etc).

Telnyx Settings

  1. Login to the Telnyx Mission Control portal.
  2. Click on the SIP Connections menu option and then Add SIP Connection.
  3. Enter a Name and select Type: IP address.
  4. Under Expert IP Auth Settings choose the option Token and register the suggested Token (or define any alphanumeric value that you prefer). Click Save.
  5. On the same connection, click + Add IP to add the following Signaling IP addresses that belong to Plivo:

    • 52.9.254.127
    • 52.9.254.123
    • 107.20.176.37
    • 184.169.138.133
    • 107.20.251.237
    • 54.215.5.82
  6. Click on Save all Changes. Your SIP Connection should look like the image below.

    Plivo

  7. Click on the Outbound menu option and create a new Outbound Profile or use an existing one.

  8. Assign the new SIP Connection to the Outbound Profile to allow outbound calls to be authorized.

Your Telnyx SIP Connection is now ready to accept traffic from your Plivo account!

Plivo Settings

Note: the following example uses a Plivo XML based application, but the same analogy and guidance for the Telnyx Token configuration is the same for any other Plivo API based application, i.e. SDK, Python, Node, etc.

  1. Login to the Plivo Portal.
  2. Create a Plivo application that points to your HTTP endpoint that will host your XML application:

    Plivo Setup

  3. Create an XML application that forwards calls to Telnyx and adds a Token header like the one from this example:

    <?xml
    version="1.0"
    ?>
    <Response>
      <Speak>Please hold while we connect the call.</Speak>
      <Dial>
        <User sipHeaders="Telnyx-Token=<your-connection-token>">
              sip:<your-destination-number>@sip.telnyx.com
        </User>
      </Dial>
    </Response>
  4. You can trigger your application by adding it to a call flow, e.g. assigning it to a Plivo phone number and make a call to that number.

  5. Once the application reaches that step, Plivo will send a SIP INVITE message to Telnyx to establish the call like the one from this example (please note the custom X-PH header for Telnyx-Token).

    INVITE sip:<your-to-number>@sip.telnyx.com SIP/2.0
    Record-Route: <sip:107.20.251.237:5060;lr;ftag=--------
    Via: SIP/2.0/UDP 107.20.251.237:5060;branch=----------
    Via: SIP/2.0/UDP 3.93.158.143:5080;received=3.93.158.143;rport=5080;branch=z9hG4bKev8HypZm2DH6g
    Max-Forwards: 39
    From: "<your-from-number>" <sip:<your-from-number>@3.93.158.143>;tag=----------
    To: <sip:<your-to-number>@sip.telnyx.com>
    Call-ID: ---------------------------
    CSeq: 11074464 INVITE
    Contact: <sip:[email protected]:5080>
    User-Agent: Plivo
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
    Supported: path, replaces
    Allow-Events: talk, hold, conference, refer
    Privacy: none
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 371
    X-SipUserAgent: None
    X-Plivo-ParentAuthID: <your-plivo-id>
    X-Plivo-Webrtc: false
    X-ParentCallUUID: ---------------
    X-PH-Telnyx-Token: <your-token>
    P-Asserted-Identity: "<your-from-number>" <sip:<your-from-number>@3.93.158.143>
    Remote-Party-ID: "<your-from-number>" <sip:<your-from-number>@3.93.158.143>;party=calling;screen=yes;privacy=off
    
    v=0
    o=Plivo 1571228231 1571228232 IN IP4 3.93.158.143
    s=Plivo
    c=IN IP4 3.93.158.143
    t=0 0
    m=audio 28282 RTP/AVP 0 8 9 102 3 101t
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:102 SPEEX/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtcp:28283 IN IP4 3.93.158.143
    a=silenceSupp:off - - - -
    a=ptime:20
  6. Telnyx will accept that SIP INVITE and will place/redirect the call to the PSTN number placed in the SIP URI.

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