Adding a SIP-IN line

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<p id="docs-hc-snippet" style="display: none;">Session Initiation Protocol (SIP) technology allows voice calls and other communication services to be carried over the internet.</p>
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<p class="p">Session Initiation Protocol (SIP) technology allows voice calls and other communication services
to be carried over the internet.</p>
<p class="p">You can use SIP-IN for receiving calls from the internet. SIP-OUT, for making outbound
calls on the internet, isn't currently supported.</p>
<p class="p">For information about how to add a standard Public Switched Telephone Network
(PSTN) phone number, see <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408824192026" target="_blank">Adding phone numbers</a>.</p>
<p class="p">This article contains the following topics:</p>
<ul class="ul" id="Talk_adding_sipin_line__ul_rkm_sjx_jdc">
<li class="li"><a class="xref" href="#topic_xtn_my1_qdc">About SIP-IN lines</a></li>
<li class="li"><a class="xref" href="#topic_idv_2dt_kdc">Understanding SIP-IN technical requirements</a></li>
<li class="li"><a class="xref" href="#topic_lmp_tjx_jdc">Understanding SIP-IN line limitations</a></li>
<li class="li"><a class="xref" href="#topic_v21_tjx_jdc">Adding a SIP-IN line</a></li>
<li class="li"><a class="xref" href="#topic_i54_tjx_jdc">Deleting SIP-IN lines</a></li>
</ul>
</div>
<div aria-labelledby="ariaid-title2" class="topic nested1" id="topic_xtn_my1_qdc">
<h2 class="title topictitle2" id="ariaid-title2">About SIP-IN lines</h2>
<div class="body">
<p class="p">When you add a SIP-IN line, you can accomplish the following:</p>
<ul class="ul" id="topic_xtn_my1_qdc__ul_xmh_cz1_qdc">
<li class="li">
<strong class="ph b">Call escalation from an AI agent</strong>: Connect a third-party AI agent to
escalate calls to Zendesk agents when the AI agent cannot resolve the customer's
query. See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/8397061704474" target="_blank">Connecting an external AI agent using a
SIP-IN line</a>.</li>
<li class="li">
<strong class="ph b">Bring Your Own Carrier (BYOC)</strong>: Use local carriers and forward calls to
Zendesk via SIP to operate your business in that country.</li>
<li class="li">
<strong class="ph b">Call forwarding:</strong> Forward calls into Zendesk via SIP-IN without
paying PSTN rates for call forwarding.</li>
<li class="li">
<strong class="ph b">Third-party application (external IVR) forwarding</strong>: Connect with partner
apps during the call flow, such as an external IVR, for a more complex and
integrated service experience.</li>
</ul>
</div>
</div>
<div aria-labelledby="ariaid-title3" class="topic nested1" id="topic_idv_2dt_kdc">
<h2 class="title topictitle2" id="ariaid-title3">Understanding SIP-IN technical requirements</h2>
<div class="body">
<p class="p">Use the following information to help you understand the technical requirements for
SIP-IN lines:</p>
<ul class="ul" id="topic_idv_2dt_kdc__ul_bsg_xdt_kdc">
<li class="li">An AI agent must be able to place a call to a SIP-IN line address, (for example,
<code class="ph codeph">anyuser@department-company-zendesk.sip.twilio.com</code>). Zendesk
Talk uses Twilio's SIP service. Any user can be dialed on the SIP domain defined
in Admin Center.</li>
<li class="li">The phone number passed in the SIP URI <code class="ph codeph">From</code> header field is
used to associate an existing user profile with the ticket. </li>
<li class="li">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.</li>
<li class="li">SIP-IN lines are compatible with <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4409149119514" target="_blank">omnichannel routing</a>. When a ticket is
generated, it's routed to agents based on their availability and capacity if
you've created omnichannel routing rules.</li>
<li class="li">When creating a ticket to associate to a SIP call, you must set
the <code class="ph codeph">via_id</code> to 34 as described <a class="xref" href="https://developer.zendesk.com/documentation/ticketing/reference-guides/via-types/#:~:text=34,Phone%20call%20(incoming)" target="_blank">in the Zendesk developer documentation</a>. This identifies it as an
incoming voice call ticket in Zendesk Support. Then, when making the call,
include the SIP header: <code class="ph codeph">X-Zendesk-Ticket-Id=&lt;<em class="ph i">ticket
number</em>&gt;</code>.</li>
<li class="li">Twilio voice media IPs use a single global range: 168.86.128.0/18 with a UDP
port range 10000-60000</li>
<li class="li">Twilio supports G.711 μ-law (PCMU) and A-law (PCMA) codecs for media.</li>
<li class="li">SIP-IN lines work with your interactive voice response (IVR) trees. See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408885628698" target="_blank">Routing incoming calls with IVR</a>.</li>
<li class="li">SIP-IN lines are compatible with <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408884087706" target="_blank">customer callback</a>.</li>
<li class="li">SIP-IN lines use TLS for SIP signaling and SRTP for media, if you turn on these
secure protocols in SIP line settings.<ul class="ul" id="topic_idv_2dt_kdc__ul_cgr_45c_rhc">
<li class="li">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 <a class="xref" href="https://docs-resources.prod.twilio.com/documents/ca-bundle-sip.crt" target="_blank">Twilio's bundle of trusted CA
certificates</a>.</li>
</ul>
</li>
<li class="li">TLS support:<ul class="ul" id="topic_idv_2dt_kdc__ul_cyg_r3y_4hc">
<li class="li">
<p class="p">TLS 1.2+</p>
</li>
<li class="li">
<p class="p">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.</p>
</li>
</ul>
</li>
<li class="li">And, SRTP support:<ul class="ul" id="topic_idv_2dt_kdc__ul_w1m_r3y_4hc">
<li class="li">
<p class="p">Inbound to Zendesk (origination): AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32</p>
</li>
<li class="li">
<p class="p">Outbound from Zendesk (termination): AES_CM_128_HMAC_SHA1_80</p>
</li>
<li class="li">
<p class="p">Master Key Identifier (MKI) isn’t supported.</p>
</li>
</ul>
</li>
</ul>
</div>
</div>
<div aria-labelledby="ariaid-title4" class="topic nested1" id="topic_lmp_tjx_jdc">
<h2 class="title topictitle2" id="ariaid-title4">Understanding SIP-IN line limitations</h2>
<div class="body">
<p class="p">The following limitations currently apply when you create SIP-IN lines:</p>
<ul class="ul" id="topic_lmp_tjx_jdc__ul_nwd_cjt_kdc">
<li class="li">Only SIP-IN is available for this release; SIP-OUT isn't currently supported for
outbound calls.</li>
<li class="li">
<a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408881907994#topic_fkr_j2d_yt" target="_blank">Failover functionality</a> isn't supported
on a SIP-IN line.</li>
<li class="li">SIP-IN lines cannot be configured using APIs. They can only be set up
in Admin Center.</li>
<li class="li">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.</li>
<li class="li">You can’t <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408834961050" target="_blank">block numbers</a> from a SIP-IN line.</li>
<li class="li">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.</li>
</ul>
</div>
</div>
<div aria-labelledby="ariaid-title5" class="topic nested1" id="topic_v21_tjx_jdc">
<h2 class="title topictitle2" id="ariaid-title5">Adding a SIP-IN line</h2>
<div class="body">
<p class="p">Talk admins can create SIP-IN lines.</p>
<p class="p"><strong class="ph b">To add a SIP-IN line</strong></p>
<ol class="ol" id="topic_v21_tjx_jdc__ol_jjr_gkx_jdc">
<li class="li"><span class="ph">In <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4581766374554#topic_hfg_dyz_1hb" target="_blank">Admin Center</a>, click <img class="image" id="topic_v21_tjx_jdc__d8e981" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/ac_channels_icon.png">
<strong class="ph b">Channels</strong> in the sidebar, then select <strong class="ph b">Talk and email &gt; Talk</strong>.</span></li>
<li class="li">Click the <strong class="ph b">Lines</strong> tab, where you can review your existing phone numbers,
digital lines, and SIP lines and add new ones.</li>
<li class="li">Click <strong class="ph b">Add Line</strong> &gt; <strong class="ph b">Add SIP line</strong>.<p class="p"><img class="image" id="topic_v21_tjx_jdc__image_rcl_4fz_mdc" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/Talk_adding_sipin_line22.png" width="250"></p>
</li>
<li class="li">On the SIP configuration tab, configure the following settings:<ul class="ul" id="topic_v21_tjx_jdc__ul_ijx_skx_jdc">
<li class="li">
<strong class="ph b">Nickname</strong>: 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.</li>
<li class="li">
<strong class="ph b">URI</strong>: Enter a unique SIP domain prefix. Zendesk appends
<code class="ph codeph">-zendesk.sip.twilio.com</code> to form your full SIP
domain. For example:
<code class="ph codeph">department-company-zendesk.sip.twilio.com</code>.<p class="p">When
making a SIP call, you use the domain in the SIP URI. For example:
<code class="ph codeph">anyuser@department-company-zendesk.sip.twilio.com</code>.</p>
</li>
<li class="li">
<strong class="ph b">Secure media (TLS/SRTP)</strong>: 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.</li>
<li class="li">
<strong class="ph b">Authentication</strong>: Click <strong class="ph b">Add IP address range</strong> 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 <strong class="ph b">32</strong> as the CIDR prefix length from
the menu.</li>
</ul>
</li>
<li class="li">On each tab of the line settings page, configure the settings you want
using the following articles as a reference:<ul class="ul" id="topic_v21_tjx_jdc__ul_bz1_p1t_kdc">
<li class="li">
<strong class="ph b">Settings tab</strong>: See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408823877146" target="_blank">Managing line settings</a>.</li>
<li class="li">
<strong class="ph b">Voicemail tab</strong>: See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408831899930" target="_blank">Configuring voicemail
options</a>.</li>
<li class="li">
<strong class="ph b">Routing tab</strong>: See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408885952922" target="_blank">Routing incoming calls to groups of
agents</a>.</li>
<li class="li">
<strong class="ph b">Call recording tab</strong>: See <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4408831738266" target="_blank">Managing call recording
options</a>.</li>
</ul>
<div class="note note">
<span class="notetitle">Note:</span> Not all of the settings will be available for SIP lines.</div>
</li>
<li class="li">When you're finished, click <strong class="ph b">Save</strong>.</li>
</ol>
<p class="p">Your SIP-IN line is now set up.</p>
</div>
</div>
<div aria-labelledby="ariaid-title6" class="topic nested1" id="topic_ktm_g2c_qdc">
<h2 class="title topictitle2" id="ariaid-title6">Editing SIP-IN lines</h2>
<div class="body">
<p class="p">After you’ve created a SIP line, you can edit its settings if you want to
change anything.</p>
<p class="p"><strong class="ph b">To edit a SIP line</strong></p>
<ol class="ol" id="topic_ktm_g2c_qdc__ol_ltm_g2c_qdc">
<li class="li"><span class="ph">In <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4581766374554#topic_hfg_dyz_1hb" target="_blank">Admin Center</a>, click <img class="image" id="topic_ktm_g2c_qdc__d8e981" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/ac_channels_icon.png">
<strong class="ph b">Channels</strong> in the sidebar, then select <strong class="ph b">Talk and email &gt; Talk</strong>.</span></li>
<li class="li">From the list of SIP lines, click the gear icon (<img class="image" id="topic_ktm_g2c_qdc__image_w3g_hvy_4hc" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/aiaa_side_settings_icon.png" width="16">) next to the SIP-IN line you want to
edit, and then select <strong class="ph b">Edit</strong>.<p class="p"><img class="image" id="topic_ktm_g2c_qdc__image_otm_g2c_qdc" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/Talk_adding_sipin_line36.png" width="550"></p>
</li>
<li class="li">Edit the settings as required using the instructions in the <a class="xref" href="#topic_v21_tjx_jdc">Adding a SIP-IN line</a>
section.</li>
<li class="li">Click <strong class="ph b">Save changes</strong>.</li>
</ol>
</div>
</div>
<div aria-labelledby="ariaid-title7" class="topic nested1" id="topic_i54_tjx_jdc">
<h2 class="title topictitle2" id="ariaid-title7">Deleting SIP-IN lines</h2>
<div class="body">
<p class="p">After you’ve created a SIP line, you can delete it if you no longer need
it.</p>
<p class="p"><strong class="ph b">To delete a SIP line</strong></p>
<ol class="ol" id="topic_i54_tjx_jdc__ol_pwb_bbt_kdc">
<li class="li"><span class="ph">In <a class="xref" href="https://support.zendesk.com/hc/en-us/articles/4581766374554#topic_hfg_dyz_1hb" target="_blank">Admin Center</a>, click <img class="image" id="topic_i54_tjx_jdc__d8e981" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/ac_channels_icon.png">
<strong class="ph b">Channels</strong> in the sidebar, then select <strong class="ph b">Talk and email &gt; Talk</strong>.</span></li>
<li class="li">From the list of SIP lines, click the gear icon (<img class="image" id="topic_i54_tjx_jdc__image_urx_3vy_4hc" src="https://zen-marketing-documentation.s3.amazonaws.com/docs/en/aiaa_side_settings_icon.png" width="16">) next to the SIP line you want to
delete, and then select <strong class="ph b">Delete</strong>.</li>
<li class="li">Click <strong class="ph b">Delete SIP line</strong> to confirm you want to delete the SIP
line.</li>
</ol>
</div>
</div>
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