The Shout SIP-Switch has been designed from the ground up by Shout Telecoms engineers using standard server technology and robust design. The SIP switch is a highly flexible, secure and fully featured platform with multiple service creation options.
The Shout SIP-Switch terminates SIP trunking and can provide call-routing decisions under the control of the onboard Application Server, or for larger deployments we have an API for external control.
The SIP-Switch can also provide audio playback and DTMF detection capability on-switch, thus reducing the need for external IVR support for audio announcements and menu selections.
Network Address Translation (NAT)
Support of near-end and far-end NAT to allow NAT traversal of signalling and media. Near-end NAT allows the SIP Technology Platform to be located behind a router or border function whilst far-end NAT allows detection of SIP Trunks and User Agents behind a router. This combination of near-end and far-end NAT allows for flexible deployment options.
Call routing is under Application Server control allowing fully programmable routing and dial plans. Our Application Server supports real-time database lookups and web service APIs for powerful custom developed applications with 3rd party integration. Support for SIP Redirect for efficient redirect or number translation applications under Application Server control.
3rd Party Call Control and Multiple Leg Support
Support for multiple legs and switching of media streams between legs under Application Server control allowing multiple leg applications such as call agent applications and network IVR announcements.
Back-to-Back User Agent (B2BUA) Architecture
The SIP Technology Platform B2BUA Architecture allows the SIP Technology Platform to terminate a SIP Trunk and create new outbound call legs with modified SIP headers. This differs from SIP Proxy behavior which only allows SIP messages to be re-routed at an IP level. B2BUA architecture allows complete control of the outbound SIP Headers allowing for Session Border Controller functionality, NAT Traversal and privacy/security.
Audio Routing and Media Server Functionality
- RTP/Media Stream options
- Per SIP Trunk options for; signalling only (RTP direct from trunk-to-trunk for efficiency), RTP pass through SIP Technology Platform (for NAT traversal) or an RTP end-point on SIP Technology Platform (for audio play or DTMF detection on SIP Technology Platform).
- Audio Play
- Audio play on all legs; supports A-party audio for IVR/Media Server functionality or network whisper messages to A and B parties. Instant audio deployment for flexible service creation.
- DTMF Detection
- Support for SIP INFO and RFC2833 DTMF detection. DTMF can be detected whilst playing audio prompts or enabled mid call to detect DTMF events to signal to the Application Server.
- Audio Record
- Record audio on any call leg, then FTP or Email the recording to an external destination.
- Conversation Record
- Record a 2-Party conversation as a stereo file with options to encrypt recordings.
Session Management User Interface
- Web based user interface administration
- Monitor calls in progress, call details and force clear-downs
- SIP messaging and call state trace debug with filters
- Call statistics SIP Trunk and Trunk Group administration
- Session Management information available via SOAP API
Black/White list of SIP Trunk IP addresses
SIP Trunk Groups/Routes
- Independent Inbound and Outbound SIP Trunks
- Application/Service selection from SIP Trunk Group
- Call count optionally limited per SIP Trunk Group to manage inbound and outbound SIP Trunk allowances
- Fail-over and load balance across SIP Trunks in Trunk Group
Call Detail Records
CSV and Telsis Binary format - stored on-disc for FTP retrieval
SIP UDP RFC3261
Performance and Licence Model
Performance – High performance thread pool architecture. Calls/Second and BHCA is application and hardware specific, typical calls/second for switched calls are 50 calls/second under Application Server control, per unit.
Licences – The SIP Technology Platform is licensed based on inbound calls and switched call legs. Additional licences are available for audio play and DTMF detection. The SIP Technology Platform can be licensed for up to 960 switched call per unit. Multiple units can be clustered and remotely sited with purchased licences automatically pooled across the entire estate.
- Fully Programmable
- Comprehensive API’s for web portal integration
- Simple Service Creation scripting Language (training provided)
- Web Management Portals are also available
- Shout service creation. We frequently develop, integrate and support new voice applications for our customers
Class 4 SIP Switch
- For core network point to point trunking
- Carrier-grade: approved for BT IP Exchange, Cable and Wireless, plus proven connectivity to many other SIP interconnects.
Class 4/5 Hybrid Switch
- For core network trunking with intelligent voice and call control applications
- Managing SIP traffic behind an SBC, adding intelligent call control to a network and optimising network utilisation
Delivering a comprehensive range of voice applications, including:
- DTMF detection
- Audio play
- Audio record
- Audio play speed/skip via DTMF
- Voice Activity Detection
- Text To Speech
- Speech Recognition
- Call queuing
- Voice menus
- Email and FTP support for end-to-end applications
Voice Recording Server
- In-Line (tromboned)
- RTP sniffer or port mirrored
- Supports Wav and MP3
- Connect securely direct to IP phones
- Multiple and distributed architecture for high call volumes and resilience
- Automatic failover and replication ensure high availability
- Master/slave, load-share and N+1 redundancy configurations
- Every first purchase comes with a free secondary unit for redundancy
- Licences pooled across the entire estate ensuring capacity is never lost
- Web managed and SNMP support
- Graceful shutdown for managed upgrades and system maintenance
- Automatic system restart in the event of application failure
- Suite of application management and monitoring tools
Backed by our 24/7 Global support.