We're excited to announce the release of SIP-IN (Session Initiation Protocol) integration for Zendesk Talk, designed to enhance your communication experience by integrating the Talk call flow with external carriers, known as Bring Your Own Carrier (BYOC) and applications.
Main documents are now online:
SIP helps connect with external partners during the call flows enabling real-time communication between multiple parties, including agents, customers, and partners, for efficient collaboration and decision-making.
Example Use Cases solved by SIP:
- Call escalation from a virtual agent: As a customer, I want to connect a third party virtual agent with Zendesk Voice and escalate calls to Zendesk agents in cases where the virtual agent is not able to resolve my query.
- BYOC: As a customer, I need to use local carriers and forward calls into Zendesk via SIP so that I can operate my business in that country.
- Call Forwarding: As a customer, I want to forward calls into Zendesk via SIP, so that I don’t have to pay PSTN rates for call forwarding.
- 3rd Party Application (external IVR) Forwarding: As a customer, I want the ability to connect with partner apps during the call flow, such as an external IVR, to enable a more complex and integrated service experience.
- Any other cases requiring call forward to Zendesk agents
Call Flow:
What does the experience look like?
Admin Experience (Settings):
Customers can create a SIP line similar to a PSTN or digital phone line in Zendesk Voice Admin settings. This allows them to configure various settings, including SIP connection details (setting up SIP domain and authentication) and other line administration settings such as greetings, routing etc..
Here is the glimpse of the experience:
Requirements for SIP-IN Calls into Zendesk Voice:
- Third parties should be able to place a call on SIP domain address (e.g., mongoose@mongoose.sip.twilio.com ) configured in Zendesk Admin Settings. Zendesk uses Twilio’s SIP in the background. Any user can be dialed on the defined SIP domain address.
- Authentication Mechanism: Zendesk employs IP access control lists. IP ranges from third parties can be defined in Zendesk Admin settings. We do not employ any other authentication method such as username and password.
- The Secure Real-time Transport Protocol (SRTP) is not supported over the SIP; only the Real-time Transport Protocol (RTP) is supported at this time.
- Medic Codec and IP allowlist -
- Our provider supports G.711 μ-law (PCMU) and A-law (PCMA) codecs for media.
- Twilio Voice Media IPs will use a single global range; 168.86.128.0/18 with a UDP port range 10000-60000
- [Optional] A ticket ID can be passed to Zendesk to be associated with the SIP call via SIP headers.
- The SIP header format should be: X-Zendesk-Ticket-Id=1217
- When creating the ticket, please use the via_id 34 as detailed [here]. This will help identify it as a voice incoming call ticket in Zendesk.
- Additional connection details will be shared soon.
- The SIP line will be compatible with omnichannel routing.
Agent Experience:
- Agents will be shown the linked call tickets containing caller and SIP details when handling SIP-IN calls, ensuring a seamless conversation.
- The phone number passed in the SIP URI “from” will be used to associate any existing user profile with the ticket. (+1-xxx-xxxx@mongoose.sip.twilio.com)
Reporting and Billing:
- Historical Reporting: Admins can generate historical reports via incremental APIs to analyze call center performance.
- Real-time Monitoring: SIP-IN call data is available on the real-time Voice dashboard for instant performance analysis.
- SIP calls will be charged per minute basis as other calls in Zendesk. The exact per minute amount to be confirmed.
- There will be no costs for adding a SIP line.
- Billing Integration: Payment for SIP calls is integrated into the current credit usage system, with detailed charge breakdowns available on the Usage charges page for reconciliation.
LIMITATIONS:
Please leave any feedback for any of the limitations mentioned, your feedback will be considered in prioritizing the feature requests.
- Only SIP-IN is available for this release; SIP-OUT is not yet supported. We will keep improving our initial release and re-solving the limitations.
- Authentication Mechanism: IP access control lists are used for authenticating SIP requests on the Zendesk Voice SIP line. Other mechanisms, such as username and password, are not yet supported.
- The Secure Real-time Transport Protocol (SRTP) is not supported over the SIP; only the Real-time Transport Protocol (RTP) is supported at this time.
- Configuration: SIP lines cannot be configured via APIs and can only be set up in the Zendesk Admin Center.
- Callback Functionality: Callback functionality is not supported on a SIP line.
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Outbound Calls: There is no ability to place outbound calls from a SIP line, except in cases of call overflow and agent forwarding, where customers will need to choose another PSTN number in Zendesk for these purposes.
- Workaround: An external number can be added for outbound calling.
- Bring Your Own Carrier (BYOC): The ability for a carrier to integrate with Zendesk SIP may depend on implementation specifics of SIP. Carriers might need time for testing and integration. This release might not be compatible with every carrier/partner. Please highlight the connection details to the carrier/partner you are working with.
- Blocked number functionality does not work for a SIP line.
- The admin screens are only available in English at the moment.
- Any more limitations will be listed here as we develop the functionality.
How much does a SIP call cost?
- SIP calls will be charged per minute basis as other calls in Zendesk. $0.01/min
- Per minute price does not vary with geography
- There is no subscription cost for setting up a SIP Line
- Bill comes as part of Zendesk Voice usage
- Charges corresponding to each leg can be reconciled using Talk Usage charges page
Who is this available for?
SIP lines will be available to all Zendesk suite customers who use Voice.
Release timeline and feedback
Rollout: Commencing 9th Dec, 2024 - Concluding 16th Dec, 2024.
