SIP Registration
Manage SIP user registrations and authentication workflows across supported endpoints.
SIPUI SwiTch brings together retail calling cards, wholesale IP-to-IP routes, reseller billing, SIP registration, PBXUI integration and AI-enabled RTP media workflows in a flexible server-based communication solution. Deploy on your own Linux server or explore a managed hosted plan.
Building blocks for managing the SIP signaling, routing and operational side of your communication platform.
Manage SIP user registrations and authentication workflows across supported endpoints.
Configure destinations, priorities, route selection and failover for outbound traffic.
Connect call detail records, rating and account balance workflows to a compatible billing system.
Support signaling and call lifecycle handling for SIP-based voice services.
Server-focused deployment with installation and configuration guidance for supported Linux environments.
Choose a hosted environment or discuss a deployment on infrastructure you control.
View permission-based, time-stamped location records for authorized users and devices, with date and time filtering, chronological history and interactive map view where location data is available.
Plan and manage TCP/UDP service ports and NAT port-forwarding rules for SIP signaling, RTP media and supported services, subject to server and network access.
Deploy an SBC at the SIP network edge for controlled trunk interconnection, SIP signaling inspection, topology hiding and RTP media policy, subject to configured modules.
IP-based allow/deny rules, trusted SIP endpoints, address restrictions and review of firewall policies for supported Linux deployments.
Plan certificate installation, renewal and encrypted management or SIP signaling endpoints where supported by the selected deployment.
Integrate SIP trunks, extension management, call routing and supported PBXUI administration workflows subject to compatibility testing.
Connect compatible AI speech services for voice-assistant workflows, transcription, call automation and event-driven PBX integrations.
Record authorized RTP audio streams with call metadata, optional per-leg recording and configurable retention policies.
Convert decoded RTP speech into searchable text using supported speech-to-text processing, with optional real-time transcripts.
Turn text into synthesized speech, encode to a negotiated audio codec and deliver audio on an established RTP session.
Calling card account and PIN workflows, rate plans, customer balance tracking, access numbers and retail call records for supported configurations.
Route wholesale traffic between authorized SIP interconnects using IP authentication, trunk configuration, routing priorities and failover.
Architectures targeting up to 50,000 simultaneous calls across properly sized infrastructure; actual capacity requires benchmarking and deployment validation.
Create multi-level reseller and sub-reseller structures without a predefined application-level tier cap; actual operational limits depend on the chosen license and resources.
Reseller branding, customer rate plans, balance management, usage reports, CDR rating and invoice workflows subject to enabled billing modules.
From requirements to launch, the solution is scoped for your traffic and operational needs.
Select a hosted Linux server plan or request self-hosted deployment.
Configure SIP ports, users, codecs, routing and security settings.
Add SIP trunk connections and define routing and billing integrations as needed.
Validate registration, calling, reporting and monitoring before production rollout.
Plan Linux administration, multiple audio/video codecs, media and data limits, low-bandwidth policies, modular integrations and multi-location coordination from a central management zone. Capabilities depend on deployed modules, access privileges and network architecture.
Configure SIP UDP/TCP ports, RTP media ranges, allowed network interfaces and NAT forwarding requirements. Changes should be tested to prevent disrupting active calls.
Use allowlists and blocklists for trusted IP addresses and CIDR ranges, with rule review and protection for administration access.
Set up certificates for supported HTTPS and TLS endpoints; review expiry dates and renewal procedures.
Connect compatible SIP and PBX administration workflows, subject to version compatibility and integration testing.
Extend supported deployments with modular SIP, routing, billing, reporting, media processing, monitoring and third-party integration components. Module compatibility is validated per release.
Organize servers across regional sites and data centers with location-specific status, roles, routing configuration and operational visibility.
Use a central management zone to coordinate authorized server inventory, configuration distribution, health checks and operational policies across locations.
Plan inter-region routing, route priorities and failover strategies while accounting for WAN connectivity, latency and location-specific capacity.
Aggregate server health, registration metrics, alarms and selected audit events from connected locations with role-based access controls.
Allow the same authorized SIP user identity to register through different regional SIPUI SwiTch servers using a centrally coordinated PHP API. Each server maintains its own live registration contacts and expiry times while shared identity and account policies remain consistent.
Connect regional SIP registration workflows to a central PHP API for account checks, operator-code mapping, enabled status and server eligibility. Use authenticated server-to-server requests, TLS, rate limits and short-lived caching.
Integrate with a MySQL/MariaDB account database for SIP identities, operator mappings, account status and authorized server assignments. Support secure queries, indexing and controlled synchronization without storing plaintext SIP passwords.
Route SIP MESSAGE requests between authorized users on different regional servers, with Central Zone destination lookup, server-to-server delivery and per-message tracking.
Share supported photos, audio, video and documents across server locations using secure HTTPS uploads, access-controlled media links and database-backed attachment metadata.
Use SIP MESSAGE notifications or delivery events to alert recipients that new messages and shared files are available, including when the sender and recipient use different servers.
Coordinate message IDs, duplicate protection, offline queues, delivery acknowledgements and controlled retries between connected servers through authenticated PHP APIs.
Define per-operator and per-server data retention, upload permissions, bandwidth quotas, transfer logs, access rules and storage allocation from Central Zone. Apply policy by account, reseller or region.
Set configurable video upload size limits, maximum duration, resolution and bitrate; offer server-side transcoding or adaptive compression where a media-processing module is deployed.
Limit audio or voice-message file size and duration, select supported efficient codecs and bitrate targets, and enforce per-user or operator media policies without breaking SIP codec compatibility.
Configure per-file upload limits, allowed file types, authorized download URLs, quota limits and expiry rules. Validate files server-side and track upload status through MySQL-backed metadata.
Reduce bandwidth with optional media compression, thumbnail generation, small previews, download-on-demand, caching and transfer deduplication. Audio/video quality and CPU requirements may vary.
Configure audio codec priorities and SIP/SDP negotiation for supported Opus, G.729, G.711 PCMU/PCMA and other installed codec modules. Codec availability and licensing depend on the deployment.
Manage supported H.264, VP8 and VP9 video codec preferences with resolution, frame rate and bitrate profiles by endpoint, operator or server. Playback and calls depend on device and module compatibility.
Use installed media modules to convert between compatible audio/video codecs when call endpoints cannot negotiate a common format. Transcoding requires adequate server CPU and licensed codec support where applicable.
Distribute codec allowlists, codec priority, video profiles and bitrate ceilings to authorized multi-location servers; configure per-operator defaults and validate policies before rollout.
Distribute versioned, authenticated size limits, codec presets and storage policies from Central Zone to connected regional servers, with audit logs, staged rollout and safe rollback.
Protect remote administration with authenticated encrypted channels, scoped permissions, change auditing and tested rollback procedures.
Security note: firewall, SSH, NAT and certificate changes require administrator authorization and safe rollout with backup/rollback plans. Media limits and low-byte optimization require application/server enforcement and optional transcoding modules. This webpage describes solution areas only; it does not operate remote servers, authenticate SIP users, synchronize registrations, deploy modules or change Linux firewall settings. Centralized actions require an authorized controller/agent backend and secure connectivity.
Build retail prepaid services, manage wholesale SIP traffic and deliver branded reseller billing from a scalable platform.
PIN-based calling card services, customer balances, destination rates, purchase/recharge workflows and call usage visibility, according to deployment setup.
Authenticated SIP trunk peers, incoming and outgoing route management, failover, CDR generation and billing reconciliation for authorized carriers.
Branded reseller portals, independent rate decks, multi-level accounts, usage statements and configurable billing access.
Enterprise multi-node scaling objective for wholesale deployments. Actual simultaneous-call capacity depends on SIP and RTP processing, codecs, transcoding, media anchoring, CPU, network, storage and topology, and must be load-tested before any commitment.
“Unlimited reseller” describes a configurable reseller hierarchy, not unlimited free usage or guaranteed infinite server capacity. Feature availability, licensing and service terms must be confirmed for each installation.
Connect PBX call control with RTP recording, speech recognition and synthesized audio for supported SIP applications and AI-enabled call flows.
Connect supported PBXUI workflows, SIP trunks, extensions, routing, call events and compatible administration interfaces.
Integrate supported speech AI services and application logic for voice assistants, interactive menus, transcription and call insights.
Capture decoded call audio from authorized media paths, associate recordings with CDR or Call-ID and apply storage access controls.
Receive RTP, decode its negotiated codec, process audio through speech recognition and deliver transcript events to connected applications.
Generate speech from text, resample and encode for the negotiated codec, then packetize and send on an authorized RTP stream.
Architect RTP ingress, codec handling, optional AI processing and RTP egress with jitter, timing and latency considerations.
SIP Call → Negotiated RTP Audio → Decode → Record / Speech-to-Text → AI Processing → Text-to-Speech → Encode → RTP Audio Return
Availability depends on active modules, compatible codecs, processing resources and provider configuration. AI services, transcription accuracy, live latency and recording require testing. Recording and transcription must follow applicable consent and privacy rules.
Protect and interconnect SIP networks with simple, policy-based session border management for retail, wholesale, PBX and multi-location SIPUI SwiTch deployments.
Apply SIP trunk IP allowlists, per-peer authorization, connection policies and rate limits at the network edge.
Coordinate public/private address traversal and optional RTP anchoring or media relaying with suitable network and codec configuration.
Reduce exposure of internal network addresses through controlled SIP header rewriting and boundary policies.
Adapt compatible SIP headers, transport settings and trunk interconnect rules to improve network compatibility.
Configure SIP TLS and supported SRTP media policies where both peers, certificates and modules permit it.
Coordinate regional SIP edge policies, route priorities and failover configuration through an authorized Central Zone.
SBC functionality requires configured SIP signaling/media components, security rules and network testing. This page presents deployment capabilities and does not itself enable an SBC or guarantee attack prevention.
Connect users registered to different SIPUI SwiTch servers through a coordinated delivery workflow, with MySQL-backed message metadata and an authenticated Central Zone PHP API.
Look up the recipient's active server, relay SIP MESSAGE safely between trusted servers and keep a unique message ID to prevent duplicate delivery.
Upload attachments over HTTPS, store file references and size/type metadata in MySQL, and deliver an authorized download notification instead of embedding large files in SIP messages.
Notify connected SIP users about incoming messages and available attachments. Offline users can retrieve pending events after registration, subject to configured retention and device behavior.
Central Zone maps operator code, SIP identity and last authorized server; regional gateways can deliver events without duplicating account databases.
Track sender, recipient, originating server, destination server, message type, attachment identifier, status and timestamps with indexed queries.
Protect server APIs with signed requests or mutual TLS, check attachment access per recipient, and enforce rate limits, file scanning and retention rules.
Sender (BD Server) → SIP MESSAGE / File Upload → Central Zone Lookup & MySQL → Recipient (SG Server) → SIP Notification → Message / File Retrieval
SIP MESSAGE handles lightweight messages and event notifications; actual file content is transferred separately over authenticated HTTPS. This page describes a proposed integration architecture, not an active message relay or file-sharing backend.
Choose server location, concurrent call capacity and currency. Monthly prices are retrieved from the MySQL database.
Entry hosted configuration
Expanded hosted configuration
Larger / custom configurations
Pricing varies with selected region and concurrency tier in this sample calculator. Capacity is a planning target, not a guaranteed channel allowance; hardware, codec, transcoding, bandwidth, recording and call patterns affect achievable performance. Setup, taxes, maintenance, SIP minutes, trunks and service-level commitments are not included or guaranteed.
Choose a concurrent-call tier, license term and currency. Prices load from the license database API; server hosting is not included.
Loading license prices…
Sample pricing must be replaced before publishing actual rates. USD and BDT amounts are saved separately in MySQL. License keys, activation, maintenance, taxes, service limits and support terms require confirmation. Hosting and telecom usage are separate.
Choose the infrastructure model that matches your calling traffic, management preferences and security requirements.
All planning topics in six easy-to-scan groups.
Hosted VPS, dedicated servers, concurrent calls, high-volume load testing and multi-location Central Zone control.
SIP trunks, failover, port forwarding, SBC, IP-based firewall, NAT, SSL/TLS and access policies.
MySQL/PHP API integration, cross-server registration, SIP messages, offline queues, notifications and file sharing.
Audio/video codecs, PBXUI, RTP recording, transcription, speech synthesis, AI call automation and media modules.
CDR rating, retail calling cards, wholesale IP-to-IP traffic, prepaid accounts, reseller tiers and white-label billing.
Upload limits, low-byte compression, storage policies, retention, monitoring, backups, logs and maintenance.
Final features, deployment responsibilities, capacity and fees depend on selected modules, infrastructure and agreed service terms.
This page presents server-based hosted plans as example configurations. Contact the team to confirm available hosting, management responsibilities and final monthly charges.
Self-hosted deployment can be discussed. Compatibility and installation requirements depend on your Linux distribution, server resources and specific configuration.
Billing integration is part of the solution scope. Supported billing features and whether billing is included in your package must be confirmed before purchase.
No such inclusions are promised by these sample plans. Confirm SIP trunk connectivity, channel limits, traffic rates and recurring costs in the final quote.
These workflows can be scoped for supported installations. Firewall and network permissions, hosting provider restrictions and safe access to the Linux server determine what can be configured.
SSL/TLS certificate management and PBXUI integration are supported solution areas. Confirm the precise features, compatible versions and installation requirements in your quote.
These are supported deployment scenarios with corresponding billing and routing modules. Confirm the exact card authentication, recharge, SIP trunk and integration requirements before production.
No. 50K is an enterprise architecture target, not a guarantee for a single server or a default hosted plan. The required infrastructure, codecs and call path must be benchmarked and validated.
The platform can be configured for multi-level reseller hierarchies, but license conditions, physical resources and operational policies still apply. Branding, reporting and billing modules should be specified in the implementation agreement.
These are integration capabilities that require matching SIP/RTP codecs, media-processing modules, optional AI/STT/TTS providers, suitable hardware and project-specific configuration. Real-time performance and availability must be validated in your deployment.
A central control architecture can be designed to connect authorized Linux servers across regions for monitoring, coordinated configuration and operational workflows. Actual control requires deployed controller/agent services, secure connectivity and compatibility testing.
Yes, this can be designed using shared account authorization through a central PHP API and MySQL/MariaDB data. Each SIP server must manage its own REGISTER contacts and expiration; registering on multiple servers does not automatically synchronize live SIP registration state, call routing or inbound reachability. Those require explicit routing, contact aggregation and failover logic.
Integration is possible when the schema, identity format, password-verification method and access rules are compatible. The PHP API should use prepared queries, least-privilege database accounts and authenticated TLS connections between servers.
This can be implemented using SIP MESSAGE relay, central recipient lookup, authenticated server APIs, MySQL message tracking and HTTPS-based attachment delivery. Notifications can be sent through SIP, but offline delivery requires queues and client synchronization; file content should not be carried directly in large SIP messages.
No. SIP trunk permissions, signaling normalization, firewall/NAT policies, optional RTP anchoring and TLS/SRTP must be configured and verified for the selected deployment. A firewall alone is not a full session border controller.
Yes. Send your estimated concurrent calls, SIP trunk count, hosting preference and billing requirements to request a scoped proposal.
Tell us your required calling card, wholesale IP-to-IP and reseller billing features, target concurrent calls, SIP trunks, PBXUI integration, AI voice, RTP recording, speech-to-text, text-to-RTP audio, SBC policies, hosting preferences and integration requirements. We'll use those details to discuss a server-based plan.
Email: support@sipui.com ↗