Adding a SIP-IN line


Session Initiation Protocol (SIP) technology allows voice calls and other communication services to be carried over the internet.


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Session Initiation Protocol (SIP) technology allows voice calls and other communication services
to be carried over the internet.
You can use SIP-IN for receiving calls from the internet. SIP-OUT, for making outbound
calls on the internet, isn't currently supported.
For information about how to add a standard Public Switched Telephone Network
(PSTN) phone number, see Adding phone numbers.
This article contains the following topics:

About SIP-IN lines
Understanding SIP-IN technical requirements
Understanding SIP-IN line limitations
Adding a SIP-IN line
Deleting SIP-IN lines



About SIP-IN lines

When you add a SIP-IN line, you can accomplish the following:


Call escalation from an AI agent: Connect a third-party AI agent to
escalate calls to Zendesk agents when the AI agent cannot resolve the customer's
query. See Connecting an external AI agent using a
SIP-IN line.

Bring Your Own Carrier (BYOC): Use local carriers and forward calls to
Zendesk via SIP to operate your business in that country.

Call forwarding: Forward calls into Zendesk via SIP-IN without
paying PSTN rates for call forwarding.

Third-party application (external IVR) forwarding: Connect with partner
apps during the call flow, such as an external IVR, for a more complex and
integrated service experience.




Understanding SIP-IN technical requirements

Use the following information to help you understand the technical requirements for
SIP-IN lines:

An AI agent must be able to place a call to a SIP-IN line address, (for example,
anyuser@department-company-zendesk.sip.twilio.com). Zendesk
Talk uses Twilio's SIP service. Any user can be dialed on the SIP domain defined
in Admin Center.
The phone number passed in the SIP URI From header field is
used to associate an existing user profile with the ticket.
For authentication, Zendesk employs only IP access control lists. You can define
IP ranges from third parties in Admin Center. Zendesk doesn't employ any other
authentication method such as username and password to authenticate SIP-IN
calls.
SIP-IN lines are compatible with omnichannel routing. When a ticket is
generated, it's routed to agents based on their availability and capacity if
you've created omnichannel routing rules.
When creating a ticket to associate to a SIP call, you must set
the via_id to 34 as described in the Zendesk developer documentation. This identifies it as an
incoming voice call ticket in Zendesk Support. Then, when making the call,
include the SIP header: X-Zendesk-Ticket-Id=<ticket
number>.
Twilio voice media IPs use a single global range: 168.86.128.0/18 with a UDP
port range 10000-60000
Twilio supports G.711 μ-law (PCMU) and A-law (PCMA) codecs for media.
SIP-IN lines work with your interactive voice response (IVR) trees. See Routing incoming calls with IVR.
SIP-IN lines are compatible with customer callback.
SIP-IN lines use TLS for SIP signaling and SRTP for media, if you turn on these
secure protocols in SIP line settings.
Devices in your network that communicate directly with Zendesk SIP must
be configured to trust Twilio's TLS/SSL Certificate, as Twilio is our
service provider. Use Twilio's bundle of trusted CA
certificates.


TLS support:

TLS 1.2+


Ciphers: ECDHE-RSA-AES128-GCM-SHA256, ECDHE-RSA-AES128-SHA256,
ECDHE-RSA-AES256-GCM-SHA384, ECDHE-RSA-AES256-SHA384,
AES128-GCM-SHA256, AES128-SHA256, AES128-SHA, AES256-GCM-SHA384,
AES256-SHA256, AES256-SHA.



And, SRTP support:

Inbound to Zendesk (origination): AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32


Outbound from Zendesk (termination): AES_CM_128_HMAC_SHA1_80


Master Key Identifier (MKI) isn’t supported.







Understanding SIP-IN line limitations

The following limitations currently apply when you create SIP-IN lines:

Only SIP-IN is available for this release; SIP-OUT isn't currently supported for
outbound calls.

Failover functionality isn't supported
on a SIP-IN line.
SIP-IN lines cannot be configured using APIs. They can only be set up
in Admin Center.
You can’t place outbound calls from a SIP-IN line. In the case of call
overflow and agent forwarding, you’ll need to choose an external PSTN number for
outbound calls.
You can’t block numbers from a SIP-IN line.
SIP requests originating from external Twilio accounts using TwiML (Twilio
Markup Language) to initiate SIP calls aren't supported. This includes scenarios
such as calls initiated from a Twilio IVR.




Adding a SIP-IN line

Talk admins can create SIP-IN lines.
To add a SIP-IN line

In Admin Center, click
Channels in the sidebar, then select Talk and email > Talk.
Click the Lines tab, where you can review your existing phone numbers,
digital lines, and SIP lines and add new ones.
Click Add Line > Add SIP line.

On the SIP configuration tab, configure the following settings:

Nickname: Enter a unique name for the SIP line. You'll
need this to identify the line in reports, integrate it with your apps,
and let your agents know the SIP line they are being called on.

URI: Enter a unique SIP domain prefix. Zendesk appends
-zendesk.sip.twilio.com to form your full SIP
domain. For example:
department-company-zendesk.sip.twilio.com. When
making a SIP call, you use the domain in the SIP URI. For example:
anyuser@department-company-zendesk.sip.twilio.com.


Secure media (TLS/SRTP): Turn this on to encrypt SIP signaling
with TLS and audio with SRTP for this SIP-IN line. When turned on,
non-encrypted calls to this line are rejected. This setting is per line
and is off by default.

Authentication: Click Add IP address range to add a range
of allowed IP addresses in Classless Inter-Domain Routing (CIDR)
notation. Authentication is based on the source IP address of the SIP
request. Depending on your SIP provider, you might need to enter the
provider's source or proxy IP address. To allow a single IP address,
enter the IP address and select 32 as the CIDR prefix length from
the menu.


On each tab of the line settings page, configure the settings you want
using the following articles as a reference:

Settings tab: See Managing line settings.

Voicemail tab: See Configuring voicemail
options.

Routing tab: See Routing incoming calls to groups of
agents.

Call recording tab: See Managing call recording
options.


Note: Not all of the settings will be available for SIP lines.

When you're finished, click Save.

Your SIP-IN line is now set up.



Editing SIP-IN lines

After you’ve created a SIP line, you can edit its settings if you want to
change anything.
To edit a SIP line

In Admin Center, click
Channels in the sidebar, then select Talk and email > Talk.
From the list of SIP lines, click the gear icon ( ) next to the SIP-IN line you want to
edit, and then select Edit.

Edit the settings as required using the instructions in the Adding a SIP-IN line
section.
Click Save changes.




Deleting SIP-IN lines

After you’ve created a SIP line, you can delete it if you no longer need
it.
To delete a SIP line

In Admin Center, click
Channels in the sidebar, then select Talk and email > Talk.
From the list of SIP lines, click the gear icon ( ) next to the SIP line you want to
delete, and then select Delete.
Click Delete SIP line to confirm you want to delete the SIP
line.



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