Introduction & Prerequisites
Introduction
Want to make use of Voice AI in your automation solution without overhauling your contact center? You can. Integrating your contact center with Conversational Cloud has never been easier.
With the native integration in Conversation Builder, you can easily transfer automated calls (with voice bots)to your contact center. And with some configuration in your contact center, you can similarly transfer calls from your contact center to voice bots.
To get started with integrating your contact center, follow the guidance in this section. If you have any questions, don’t hesitate to contact your LivePerson account team. We’ll help you to configure the integration between our CPaaS (Communications Platform as a Service) provider and your contact center.
Bring your own carrier (BYOC)
LivePerson supports Bring Your Own Carrier (BYOC) for LivePerson Voice AI and automation use cases.
BYOC lets you leverage Voice AI without porting your phone numbers. You can keep all of your current investments by appropriately routing calls through LivePerson’s CPaaS. The benefits of BYOC (Bring Your Own Carrier) include:
- Flexibility to transfer calls using either E.164 or SIP protocols.
- Cost optimization through sending and receiving custom data over a SIP trunk.
- Secure and compliant communication.
- Seamless integration with multiple systems via the LivePerson platform.
Below is the high level BYOC support for LivePerson Voice AI calls.
The articles that follow detail how to set up the SIP trunk with a 3rd-party contact center. To reference more 3rd-party SIP carrier configurations, refer to https://support.telnyx.com/en/collections/3968237-telnyx-sip-trunking-configurations
Configure LivePerson
Determine the transfer type, SIP or E.164, that will be used for transferring calls between your contact center and LivePerson.
You will need to have a Conversational Cloud user that has Admin privileges in order to configure the integration between our CPaaS provider and your contact center.
To configure the integration between LivePerson’s CPaaS provider and your contact center, you need to configure one or more SIP connections that allow for secure, two-way communication between LivePerson and your contact center. A SIP connection is required if you plan to use SIP to transfer calls between your contact center and Conversational Cloud.
To configure a SIP connection, you need to access the Voice Configuration module in your Conversational Cloud account. To do this:
- Log into your Conversational Cloud account as an Admin.
- On the navigational sidebar, go to Manage -> Channels -> Channel Setup -> click Configure on the Voice tile.
- On the navigation pane, go to SIP Trunk Configuration.
In this UI, you can create and configure your SIP connections.
To create a new connection:
- Click + Add SIP Connection.
- In the window that appears, choose your SIP Connection Type, and fill out the appropriate options.
Configure SIP connection settings
| Setting | Description | Required? |
| Name | A short description of the SIP connection. | Required |
| SIP Connection Type | The authentication type of the SIP connection: Credentials, IP Address, or FQDN | Required |
| Username | The user name to be used as a part of the credentials. This must be an alphanumeric value only (no spaces or special characters) and 4 to 32 characters in length. | Required if SIP Connection Type is FQDN or Credential |
| Password | The password to be used as a part of the credentials. This must be 8 to 128 characters in length. | Required if SIP Connection Type is FQDN or Credential |
SIP connection types
| SIP Connection Type | Description |
| Credentials | Used by SIP devices that authenticate with username and passwords. SIP devices need to be registered using these credentials in order to receive inbound calls, but no registration is needed if they are intended only for making outbound calls. |
| IP Address | Recommended for SIP devices that authenticate using IP address and other advanced authentication mechanisms. This option is usually recommended if you have a static IP address that can be used. |
| FQDN (Fully Qualified Domain Name) | Recommended for SIP devices that route inbound calls to FQDNs and authenticate outbound calls using either credentials or IP addresses. |
Once you fill out your SIP Connection Type details, click Create Connection to create the SIP connection.
After the SIP connection is created, you can configure additional settings for the type of SIP connection that you selected.
Basic SIP connection settings
Use the Basic tab to configure the general settings for the SIP connection that you created. This includes settings like:
- Name of the SIP connection
- Username/password for the connection
- Webhooks
Depending on the SIP Connection Type you selected for your connection, the settings you see might differ from the image above. For more info on the various settings, see the table below.
| Setting | Description | Required? |
| Name | A short description of the SIP connection. | Required |
| SIP Connection Type | The authentication type of the SIP connection: Credentials, IP Address, or FQDN. This setting can’t be changed after you create the SIP connection. | Required |
| Username | The user name to be used as a part of the credentials. Must be 4 to32 characters in length and alphanumeric values only (no spaces or special characters). | Required if SIP Connection Type is FQDN or Credential |
| Password | The password to be used as a part of the credentials. Must be 8 to 128 characters in length. | Required if SIP Connection Type is FQDN or Credential |
| Webhook URL | This will be the primary URL webhook sent for every call that is made or received by this SIP connection. | Optional |
| Webhook Failover URL | If a webhook to the primary URL fails twice, it will be sent to this URL. | Optional |
| Webhook API version | The version of the webhook event that will be sent to the Webhook URLs. | Optional; default is API v2 |
| AnchorSite | The location of the data center where the RTP media of all calls made or received by this SIP connection is anchored. | Optional; default is Latency |
| Encode Contact Header | Encode the SIP contact header sent by Telnyx to avoid issue for NAT or ALG scenarios. | Optional |
Enable on-net T.38 passthrough
| N/A | Not supported |
| Enable Comfort Noise for Call on Hold | When this option is checked, Telnyx generates comfort noise when you place a call on hold. If this option is unchecked, you will need to generate comfort noise or on-hold music to avoid RTP timeout. | Optional |
| DTMF Type | Allows the user to specify the type of DTMF to be used on the call. RFC 2833 (Recommended) - The standards-based mechanism used to send DTMF digits in-band (RTP) that is supported by many vendors in the industry. Inband - Sending of information within the same band or channel used for data such as voice or video. This option is the most prone to issues. SIP Info - Used by SIP network elements to transmit digits out-of-band as telephone-events in a reliable manner independent of the media stream. | Optional; default is RFC 2833 |
| RTCP Report Frequency | The report frequency specifies the interval in seconds between sending RTCP packets back to the user’s media IP while the call is in progress. | Optional; default is 5 seconds |
| RTCP Port | The port that is used for RTCP. RTCP mux is multiplexing both the RTP and RTCP through a single UDP port. One of the reasons why RTCP mux is used is for simplifying NAT traversal since only a single port is used for media and control messages. | Optional; default is rtp+1 |
| RTCP Capture | Allow Telnyx to capture and store RTCP messages to build the QoS Reports of the Portal SIP Debugging tools. | Optional |
| IP Address Default Routing Method | Sequential routing requires specifying primary, secondary, and tertiary IP addresses that are attempted in that order when an inbound call is received. For Round-Robin routing, all IP addresses are attempted in aleatory fashion. In both cases, call attempts fail-over to the next IP address if the call is rejected or times out, except for SIP codes like 404, 486, 603. | Optional, default is Sequential |
| Expert IP Auth Settings | There are 3 different optional mechanisms to further secure the IP Address authentication: Tech Prefix: A 4-digit prefix that needs to be appended to the destination number. Token: A string that must be sent in a custom SIP header <X-Telnyx-Token> on the SIP INVITE message. P Charge Info: A telephone number associated with this connection must be sent in the P-Charge-Info SIP header on the SIP INVITE message. | Optional |
| FQDN Default Routing Method | Sequential routing requires specifying primary, secondary, and tertiary FQDNs that are attempted in that order when an inbound call is received. For Round-Robin routing, all FQDNs are attempted in aleatory fashion. In both cases, call attempts fail-over to the next FQDN if the call is rejected or times out, except for SIP codes like 404, 486, 603. Telnyx honors SRV records in FQDNs. | Optional, default is Sequential |
Inbound SIP connection settings
Use the Inbound tab to configure settings that are tied to the inbound calls to your SIP connection. This includes settings like:
- SIP subdomain
- Enable SIP subdomain calls
- Enable encrypted media
- SIP invite message settings
Depending on the SIP Connection Type that you selected for your connection, the settings you see might differ from the image above. For more info on the various settings, see the table below.
| Setting | Description | Required? |
| Destination Number Format (DNIS) | Defines the format of the destination number set in the RURI and the To header of the SIP INVITE message. | Optional; default is E.164 |
| Originator Number Format | Defines the format of the origination number set in the From and the P-Asserted-Identity headers of the SIP INVITE message | Optional; default is E.164/National (e.g - 10 digits) |
| No Ringback Timeout | Defines the timeout before canceling the call if no ringback (180 or 183 messages) is received. | Optional; default is 5 seconds |
| No Answer Timeout | Defines the timeout before canceling the call if the call is not answered (200 OK message). | Optional; default is 300 seconds |
| Receive SIP Subdomain Calls | Allow SIP Subdomain calls from any source or restrict them only from other connections or applications of this same account. | Optional; default is disabled |
| Encrypted Media | SRTP enables media encryption, and each setting defines what Telnyx offers in the SIP INVITE message for Inbound calls. | Optional; default is Disabled |
| Ringback Settings | Default ringback behavior is to pass the ringback messages from the B leg to the A leg without any modifications as they are received. Enable Instant Ringback forces an instant 180 Ringing SIP message on the A leg even before the B leg is created. Generate Ringback Tone forces an instant 183 Session Progress SIP message with SDP and media of a generated ringback tone on the A leg, even before the B leg is created. | Optional; default is Default |
| Offered Audio/ Video Codecs | Define a codec list that is offered by Telnyx in the SIP INVITE message for Inbound calls. | Optional; If changing must be: G711U or G711A |
| Other Settings | Additional settings that can be configured SIP INVITE message. | Optional; default is Enable SIP Compact Headers |
| SIP Transport Protocol | The protocol used for sending the SIP INVITE. | Optional; default is UDP |
| SIP Region | Select from which SIP region will the call attempts be sent to your connection. | Optional; default depends on your region |
| SIP Subdomain | Define a subdomain so that all inbound calls that use it in the SIP RURI are routed to this connection. | Optional |
Tips
- For the “Credentials” SIP Connection Type, enable the setting for Receive SIP Subdomain Calls, and set this value to From Anyone. When you go to configure your SIP URI for your Conversation Builder voice bot, use the following format for your SIP URI: <sip-connection-username>@sip.telnyx.[com,.eu,ca]
- For the other SIP Connection Types, configure the SIP Subdomain setting. When you go to configure your SIP URI for your Conversation Builder voice bot, use the following format for your SIP URI: <sip-subdomain>@sip.telnyx.[com,.eu,ca]
Outbound SIP connection settings
Use the Outbound tab to configure settings that are tied to the outbound calls to your SIP connection. This tab lets you to modify settings like:
- SIP subdomain
- Enable SIP subdomain calls
- Enable encrypted media
- SIP invite message settings
Depending on the SIP Connection Type that you selected for your connection, the settings you see may differ from the image above. For more info on the various settings, see the table below.
| Setting | Description | Required? |
| Caller ID Override | Replace the From number received on the outbound call attempt with a new number. Only Telnyx phone numbers or Verified non-Telnyx numbers are allowed. | Optional |
| Localization Country | Define the outbound dialing format for local and international calls according to each countries conventions | Optional; default depends on your region |
| Fax Settings - T.38 Re-invited Initiated by | N/A | Not supported |
| Encrypted Media | SRTP enables media encryption, and each setting defines how Telnyx handles outbound call attempts. | Optional; default is disabled |
| Ringback Settings | Default ringback behavior is to pass the ringback messages from the B leg to the A leg without any modifications as they are received. Enable Instant Ringback forces an instant 180 Ringing SIP message on the A leg even before the B leg is created. Generate Ringback Tone forces an instant 183 Session Progress SIP message with SDP and media of a generated ringback tone on the A leg, even before the B leg is created. | Optional; default is Enable Instant Ringback (180) |
Missing Something?
Check out our Developer Center for more in-depth documentation. Please share your documentation feedback with us using the feedback button. We'd be happy to hear from you.