# SIP connectivity

Inbound SIP connectivity consists of two related network flows. SIP signaling
establishes, changes, and ends the call. RTP carries the call media after the
endpoints negotiate a media path in SDP. A working SIP connection therefore
requires both signaling and media to pass between DIDWW and the receiving SIP
system.

This page explains how the connectivity components fit together. For exact
addresses, ports, protocols, codecs, DTMF methods, and encryption mechanisms,
see the [SIP connectivity reference](sip-connectivity-reference.html).

## Connectivity flow

When DIDWW receives a call to a DID number assigned to an inbound SIP trunk,
it selects the trunk destination and sends an `INVITE`. The receiving system
returns SIP responses. If the call is answered, the SDP exchange determines
the RTP media addresses and ports used for the call.

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}}%%

sequenceDiagram

   participant Caller as Caller
   participant DIDWW as DIDWW
   participant SIP as Customer SIP system
   Caller->>DIDWW: Calls the DID number
   DIDWW->>SIP: SIP INVITE
   SIP-->>DIDWW: SIP responses and SDP
   DIDWW-->>Caller: Call is answered
   DIDWW->>SIP: RTP or SRTP media
   SIP->>DIDWW: RTP or SRTP media
   DIDWW->>SIP: BYE
   SIP-->>DIDWW: 200 OK
```

## Signaling destination selection

The trunk routing method determines how DIDWW obtains the signaling
destination.

- With **Static Endpoint**, the destination comes from the user part, host,
  and optional port configured on the trunk. DIDWW originates signaling from
  the regional Static Endpoint addresses.
- With **Dynamic Registration**, the customer system sends authenticated
  `REGISTER` requests to a DIDWW registrar. DIDWW then sends calls to an
  active Contact supplied by that registration.

The methods do not run simultaneously on one trunk. For destination
construction, DNS behavior, registration maintenance, and routing-method
selection, see [SIP routing methods](sip-routing-methods.html).

## Signaling and media paths

SIP signaling and RTP media can use different IP addresses and port ranges.
Allowing the SIP `INVITE` through a firewall does not by itself establish a
working audio path. The firewall or SBC must also allow media from the DIDWW
RTP source subnets.

The SDP offer and answer advertise the media addresses and ports that each
side expects to use. NAT, private addresses, or restrictive firewall rules can
make an advertised media destination unreachable even when signaling works.
The **Force Symmetric RTP**, **Allowed RTP IP Addresses**, and **RTP Ping**
settings can help with specific media and NAT behavior. Their exact behavior
and security considerations are documented under [Media &
DTMF](sip-trunk-settings-reference.html#inbound-sip-trunks-settings-media-dtmf).

## Firewall and SBC planning

Build firewall and SBC rules around the selected routing method and the
trunk's **Preferred Server**, **Network Protocol**, and **Transport** values.

- A Static Endpoint must accept signaling from each DIDWW location that can
  deliver its calls. When **Preferred Server** is **Auto**, allow all Static
  Endpoint signaling addresses because DIDWW can use an alternative location.
- A Dynamic Registration endpoint must be able to resolve and reach its
  selected registrar and receive the related return signaling traffic.
- Both routing methods require the DIDWW RTP source subnets and RTP port range
  to be allowed for media.
- Hostnames and DNS SRV discovery require working DNS resolution from the
  systems that use them.

Use the [Firewall and SBC allowlist](sip-connectivity-reference.html#inbound-sip-connectivity-firewall) as the factual
checklist for these rules.

## Authentication direction

Authentication occurs in a different direction for each routing method.

For a Static Endpoint, authorization is optional. When enabled, DIDWW responds
to the receiving server's SIP digest challenge using the credentials stored on
the trunk.

For Dynamic Registration, the customer system authenticates its `REGISTER`
requests to DIDWW using the generated trunk credentials. Those registrations
must remain active so DIDWW has a current Contact for inbound delivery.

## Transport and media encryption

TLS and SRTP protect different parts of the call.

- **TLS** protects SIP signaling between DIDWW and the customer SIP system.
- **SRTP** protects RTP media. The selected SRTP mode determines how the media
  keys are negotiated.

Enabling one does not automatically enable the other. Encryption applies to
the DIDWW-to-customer call leg and is not maintained end to end across other
call legs. Match the trunk's transport and SRTP settings to the capabilities
of the receiving SIP system.

## Codec and DTMF interoperability

The endpoints must negotiate at least one common audio codec for media to be
established. DIDWW presents codecs in the order configured on the trunk, with
the first codec having the highest preference.

DTMF is configured independently for sending and receiving. DIDWW supports
RTP `telephone-event` and SIP `INFO` methods. Select methods compatible
with the remote system to ensure that menu selections and other keypad input
are recognized in both directions.

## Connectivity monitoring

SIP `OPTIONS` can confirm that a signaling endpoint is reachable without
placing a call. An `OPTIONS` response does not confirm that a particular
call can be routed or that media, authentication, codecs, and DTMF are
configured correctly.

For call-specific troubleshooting, use the SIP response and routing-attempt
details in [Inbound Call Logs](../../../logs-analytics/call-logs/inbound-logs.html#inbound-cdr-logs). See
[Troubleshooting](../troubleshooting.html#inbound-trunks-troubleshooting) for the cause of each
response.

## Related resources

- [SIP connectivity reference](sip-connectivity-reference.html) - Look up
  network addresses, ports, protocols, codecs, DTMF methods, and encryption.
- [SIP routing methods](sip-routing-methods.html) - Understand Static
  Endpoint and Dynamic Registration destination selection.
- [SIP trunk settings reference](sip-trunk-settings-reference.html) - Review
  every configurable trunk field and value.
- [Service failover and load balancing](call-handling-and-delivery/service-failover-and-load-balancing.html#inbound-sip-failover-load-balancing) - Understand DNS SRV, Trunk Group,
  and routing-attempt behavior.

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