Network Voice Platforms
Versatile SIP Technology
Designed from the ground up by Shout Telecoms engineers using standard server technology and robust design. A highly flexible, secure, and fully featured SIP switching platform with multiple service creation options — carrier-grade, proven, and deployable at scale.
2,000+
Switched calls per unit
20,000+
Simultaneous calls per cluster
50/s
Calls per second per unit
Class 4/5
Hybrid soft switch capable
Overview
The Shout SIP-Switch can be deployed with an on-board Application Server for call-routing decisions and service control. For larger deployments, the Application Server (SCP) can be deployed as a dedicated server to enable centralised control of multiple switches.
The SIP-Switch also provides audio playback and DTMF detection capability on-switch, reducing the need for external IVR support for audio announcements and menu selections.
Performance & Scale: Licensed for up to 2,000 switched calls per unit, with units deployable in a single cluster to handle over 20,000 simultaneous calls. Purchased licences are automatically pooled across the entire estate.
Functional Specification
Supported RFCs
RFC 3261 / 2543SIP Session Initiation Protocol
RFC 2833RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 3326 The Reason Header Field for SIP
RFC 3581 An Extension to SIP for Symmetric Response Routing
RFC 4028 Session Timers in the Session Initiation Protocol
RFC 3398 ISDN User Part (ISUP) to SIP Mapping
RFC 5589 SIP Call Control — Transfer
RFC 2976 The SIP INFO Method
Network Address Translation (NAT)
Support for near-end and far-end NAT traversal of signalling and media. Near-end NAT allows the platform to sit behind a router or border function; far-end NAT enables detection of SIP trunks and user agents behind a router. This combination allows for highly flexible deployment options.
Call Routing
Call routing is under Application Server control, allowing fully programmable routing and dial plans. The Application Server supports real-time database lookups and web service APIs for powerful custom applications with third-party integration. SIP Redirection is supported for efficient resource usage with number translation under Application Server control.
Back-to-Back User Agent (B2BUA) Architecture
The B2BUA architecture allows the platform to terminate a SIP trunk and create new outbound call legs with modified SIP headers. Unlike SIP Proxy behaviour, B2BUA provides complete control over outbound SIP headers — enabling Session Border Controller functionality, NAT traversal, and privacy/security management.
Audio Routing & Media Server
RTP/Media Stream options
Per SIP trunk options for signalling only (RTP direct trunk-to-trunk for efficiency), RTP pass-through (for NAT traversal), or RTP endpoint on the platform (for audio play or DTMF detection).
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 — detectable whilst playing audio prompts or enabled mid-call to signal events to the Application Server.
Audio Record
Record audio on any call leg, then SFTP or email the recording to an external destination.
Secure RTP
SRTP supported. 2-party conversation recording as a stereo file with optional encryption.
Session Management & Administration
- ✓ Web-based administration interface
- ✓ 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
- ✓ Allow and Block list of SIP trunk IP addresses
- ✓ Detailed call data (CDRs) stored on-disc for FTP retrieval
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
- ✓ Failover and load balance across trunks in trunk group
Switch Deployments
Class 4 SIP Switch
Core network point-to-point trunking. Carrier-grade: approved for BT IP Exchange, Cable & Wireless, and proven across many other SIP interconnects.
Class 4/5 Hybrid Switch
Core network trunking with intelligent voice and call control applications.
Redirect Server
Managing SIP traffic behind an SBC, adding intelligent call control and optimising network utilisation.
Voice Recording Server
In-line (tromboned), RTP sniffer, or port-mirrored. Supports WAV and MP3.
Media Server
DTMF detection, audio play/record, TTS, speech recognition, call queuing, voice menus, and email/FTP support.
Registration Server
Secure direct connection to IP phones. Supports authentication, call transfers, SRTP, and TLS.
STIR/SHAKEN
Deployed as a signalling overlay to provide call authentication and attestation as an OPS operator. No core network changes required — APNF-issued certificates managed by Shout.
Mécanisme d'Authentification des Numéros
Signalling-only overlay for French MAN compliance under ARCEP's Naegelen Law. Signs the SIP FROM header as an OPS with ARCEP/APNF certificates, with no changes to your core network.
Resilience
- ✓ Multiple and distributed architecture for high call volumes and resilience
- ✓ Automatic failover and replication ensure high availability
- ✓ Active/Standby, load-share, and N+1 redundancy configurations
- ✓ Licences pooled across the entire estate — capacity is never lost
- ✓ 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 — provided by the same engineers who built the platform.
Key Features
- ✓ Carrier-grade, proven at scale
- ✓ 2,000+ switched calls per unit
- ✓ 20,000+ calls per cluster
- ✓ 50 calls/second per unit
- ✓ Class 4/5 hybrid soft switch
- ✓ B2BUA architecture
- ✓ STIR/SHAKEN & MAN ready
- ✓ Full API for external control
- ✓ Licence pooling across estate
Service Creation
- ✓ Fully programmable Application Server
- ✓ Comprehensive APIs for portal integration
- ✓ Simple scripting language (training provided)
- ✓ Web management portals available
- ✓ Shout develop, integrate & support
Related Products
Interested in our SIP Switch?
Talk to our team about deployment options, licensing, and integration with your existing network.