Gateway Overview
Cisco offers a wide variety of voice and video gateways. A gateway provides interfaces that allow the Unified Communications network to communicate with an external network. Traditionally, gateways have been used to connect the IP-based Unified Communications network to legacy telephone interfaces such as the PSTN, a private branch exchange (PBX), or legacy devices such as an analog phone or fax machine. In its simplest form, a voice gateway has an IP interface and a legacy telephony interface, and the gateway translates messages between the two networks so that the two networks can communicate.
Gateway Protocols
Most Cisco gateways offer multiple deployment options and can be deployed using any one of a number of protocols. Depending on the gateway that you want to deploy, your gateway may be configurable using any of the following communication protocols:
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Media Gateway Control Protocol (MGCP)
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Skinny Call Control Policy (SCCP)
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Session Initiation Protocol (SIP)
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H.323
Vendor Interface Cards
The Vendor Interface Card (VIC) must be installed on the gateway to provide a connection interface for external networks. Most gateways offer multiple VIC options and each VIC may offer many different ports and connection types for both analog and digital connections.
Refer to your gateway documentation for the protocols, cards, and connections that are offered with your gateway.
NSE-based Fax or Modem Passthrough for Secure SIP Lines
Before Release 15SU4, Unified Communication Manager, used to strip the fmtp attribute before forwarding a fax or modem call. As a result, the Named Signaling Events (NSE) negotiation becomes incomplete, and the receiving gateway is unaware of the supported event range. This leads to interoperability issues such as, modem upspeed failures or fallback to lower transmission speeds during fax or modem relay.
From Release 15SU4 onwards, when a SIP call involves a fax or modem negotiation, the system passes fmtp parameters between endpoints, instead of stripping them. This allows proper codec negotiation between endpoints and in turn enables seamless fax or modem pass-through functionality. The fix supports both early offer and delayed offer call scenarios across SIP to SIP and cross-protocol (SCCP to SIP) communications. The following are the supported call flows:
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SIP-to-SIP calls
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SCCP-to-SIP calls
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SIP-to-SCCP calls
The feature does not support MTP and TRP call flows.

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