IMS SIP Application Server

CreaLog SIP Application Server

The CreaLog SIP Application Server (SIP AS) is built on the proven CreaLog Platform 7 which has routed billions of calls since its first deployment more than 25 years ago, and perfectly fits into your IP Multimedia Subsystem (IMS).

Unmatched Control

It gives you unmatched control over your calls using standards like RFC 4240 and RFC 5552 to intelligently screen, route, transfer, and initiate SIP calls. In addition, the SIP AS can be combined with traditional SS7 IN Servers for INAP and CAMEL.

Interworking with MRF

The SIP AS can interwork with the CreaLog SIP Media Resource Function (SIP MRF) or third-party MRFs. It is built on a proven Host-based Media Processing (HMP) engine, which supports all relevant codecs like G.711, G.722, EVS, AMR-WB, and AMR-NB. The CreaLog SIP Media Server has played, recorded, conferenced and signaled trillions of VoIP packets and gives you complete control over those packets with its VoiceXML IVR and speech application engine.

State-of-the-art conversational speech recognition (NLU) and text-to-speech

The SIP Media Server can be enhanced with state-of-the-art speech recognition and text-to-speech (speech synthesis) by supporting the MRCP standard. All you need to add is your own media application - or deploy pre-built or custom applications provided by CreaLog.

The CreaLog SIP Appliance delivers VoiceXML IVR, Speech Recognition, TTS (speech synthesis), call conferencing, Call Recording and an optional built-in TDM-to-SIP VoIP/VoLTE/VoNR gateway. It integrates rapidly with both new and existing TDM, NGN, and IMS deployments.

The CreaLog SIP AS and MRF servers fully support virtualization but can as well be deployed in a rack-mount server.

For more information download the CreaLog Telco Catalog.


SIP inbound and outbound call support, call redirection, call rejection, routing and transfers, call leg bridging, REINVITE audio re-routing, REFER support

SIP registrar server, proxy server, authentication support

Configurable SIP port range, NAT IP translation support, Multi-Ethernet card bridging support

Enhanced security with SIP TLS and SRTP

Benefits cont.

Audio prompt announcement playback, bridging, mixing, conferencing, noise removal, fixed and dynamic gain control, voice activity detection (VAD), call recording, codec transcoding, speech recognition (ASR), speech synthesis (TTS)

DTMF tone detection and generation

Turnkey server with optional built-in PSTN/SIP gateway

Supports all major W3C, IETF, ITU and ANSI standards

General requests

+49 89 324656-0

Service Hotline (24/7)

+49 89 324656-10

Please make sure you have your credentials at hand.


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