Frequently Asked Questions
50+ answers across VoIP, AI, hiring, and pricing — everything you need to know about working with CelloIP Technologies.
50+
Questions answered
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Years experience
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Clients served
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Countries reached
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VoIP Platforms
8 questions
What is the difference between Asterisk and FreeSWITCH?
Asterisk is a PBX-first telephony framework ideal for single-site and multi-tenant PBX, call center ACD, and AGI/ARI scripting. FreeSWITCH is a softswitch-first media server built for high concurrency (10,000+ simultaneous calls), WebRTC via mod_verto, and carrier-grade Class 4/5 deployments. CelloIP recommends Asterisk for PBX and SME call centers; FreeSWITCH for large-scale softswitches and WebRTC platforms.
How long does it take to build a custom Asterisk PBX?
A basic single-site Asterisk PBX with standard features (extensions, IVR, voicemail, call queues) takes 4–6 weeks. A multi-tenant enterprise deployment with predictive dialler, CRM integration, and WebRTC softphone takes 14–20 weeks. CelloIP uses agile sprints with weekly deliverables.
Can CelloIP migrate our Asterisk system from chan_sip to PJSIP?
Yes. CelloIP has migrated 30+ production Asterisk systems from deprecated chan_sip to PJSIP. The process includes full config mapping, endpoint object creation, SIP trunk re-validation, parallel testing, and a zero-downtime cutover strategy. Migration typically takes 2–4 weeks depending on deployment complexity.
What is OpenSIPS and when should I use it instead of Kamailio?
Both OpenSIPS and Kamailio are high-performance SIP proxies. OpenSIPS has more expressive scripting and richer built-in modules for complex routing. Kamailio has a larger global community and is widely used as an SBC. For MVNO and CPaaS routing at scale, CelloIP typically recommends OpenSIPS; for enterprise SBC and WebRTC gateway roles, Kamailio. We work with both.
How many concurrent calls can Asterisk handle?
Asterisk 20/21 on properly tuned hardware can handle 500–2,000+ concurrent calls depending on codec (G.711 vs Opus), media processing, and whether media proxy (rtpengine) offloads RTP. CelloIP has deployed Asterisk clusters handling 3,000+ concurrent sessions using load-balanced pools with OpenSIPS as the front-end SIP proxy.
What is a Class 5 softswitch and do I need one?
A Class 5 softswitch provides subscriber-facing telephony services — extensions, voicemail, conferencing, IVR, billing, and call routing — for end users. It differs from a Class 4 softswitch (carrier transit routing). You need a Class 5 if you are an MVNO, hosted PBX provider, or carrier offering retail VoIP services. CelloIP builds Class 5 softswitches on FreeSWITCH/FusionPBX.
Can you integrate Asterisk with our CRM system?
Yes. CelloIP integrates Asterisk with Salesforce, HubSpot, Zoho, Freshdesk, and custom CRMs via AGI, ARI REST hooks, and AMI event listeners. Integrations include click-to-call, screen pop on inbound, automatic call logging, contact matching, and disposition codes — all delivered as a custom Asterisk module or middleware service.
What is the difference between IP PBX and a softswitch?
An IP PBX (e.g., Asterisk) serves a single organisation's internal telephony — extensions, voicemail, conference rooms, IVR. A softswitch (e.g., FreeSWITCH, Class 5) serves multiple tenants or carriers, with per-subscriber billing, provisioning APIs, and carrier interconnects. CelloIP builds both.
WebRTC
6 questions
What is WebRTC and how does it connect to existing SIP infrastructure?
WebRTC (Web Real-Time Communication) enables browser and mobile apps to make voice and video calls without plugins. It connects to SIP infrastructure via a WebRTC-to-SIP gateway — typically FreeSWITCH with mod_verto or Kamailio with rtpengine. The gateway handles DTLS-SRTP to RTP transcoding and ICE/STUN/TURN for NAT traversal.
Do WebRTC calls work on mobile browsers?
Yes. WebRTC is supported natively in Chrome, Safari (iOS 11+), Firefox, and Edge on both Android and iOS. CelloIP's WebRTC applications are tested across all major mobile browsers with adaptive bitrate, codec negotiation (Opus preferred), and fallback TURN relay for restricted networks.
What is the difference between STUN, TURN, and ICE in WebRTC?
STUN (Session Traversal Utilities for NAT) helps a client discover its public IP address. TURN (Traversal Using Relays around NAT) relays media through a server when direct P2P is blocked by firewall/NAT. ICE (Interactive Connectivity Establishment) is the framework that tries STUN first, then TURN as fallback. CelloIP deploys coturn for STUN/TURN and configures ICE in all WebRTC applications.
How do you handle NAT traversal for WebRTC to SIP calls?
CelloIP uses a combination of coturn (STUN/TURN server), rtpengine (media proxy), and ICE negotiation. For enterprise deployments, we deploy TURN servers in multiple regions to minimise latency. Approximately 30% of WebRTC sessions require TURN relay; the rest connect peer-to-peer via STUN.
What codecs does WebRTC support?
WebRTC mandates Opus for audio (wideband, high quality, low latency) and VP8/VP9/H.264 for video. When connecting to legacy SIP/PSTN systems, CelloIP performs Opus-to-G.711 transcoding at the gateway layer using FreeSWITCH or a custom media proxy.
How secure is WebRTC communication?
WebRTC is encrypted by specification: signalling is secured via HTTPS/WSS, and media is encrypted using DTLS-SRTP. CelloIP also implements certificate pinning, origin validation, and optional SRTP key rotation for enterprise deployments. WebRTC is inherently more secure than unencrypted SIP/RTP.
AI & Voice Bot
6 questions
What is a Voice AI agent and how does it differ from a traditional IVR?
A Voice AI agent uses natural language understanding to hold open-ended conversations — callers speak naturally, the AI extracts intent, and the system responds or takes action without rigid menu trees. A traditional IVR uses DTMF tones and pre-recorded prompts. CelloIP's Voice AI agents handle 85%+ of tier-1 queries automatically with sub-1.5s response latency.
Which STT (speech-to-text) engines do you work with?
CelloIP works with OpenAI Whisper (self-hosted for data privacy), Deepgram Nova-2, Google Speech-to-Text, and AssemblyAI. For low-latency applications, Deepgram (200–300ms) is preferred. For highest accuracy in noisy environments, Whisper Large v3 self-hosted delivers best results.
Can you integrate GPT-4 or other LLMs into a voice bot?
Yes. CelloIP integrates GPT-4, GPT-4o, Claude 3, Llama 3 (self-hosted), and Mistral into voice bots via Asterisk AGI, FreeSWITCH ESL, or the LiveKit Agents SDK. We handle prompt engineering, conversation state management, tool/function calling (for CRM lookups, bookings), and on-premise LLM deployment for data-sensitive clients.
What is the typical latency of a CelloIP-built AI voice agent?
End-to-end latency (speech-in to speech-out) is typically 1.2–1.8 seconds: 200–400ms STT, 300–600ms LLM inference (with streaming), and 300–500ms TTS synthesis. CelloIP optimises each component — using streaming STT, streaming TTS, and parallel LLM prefill — to achieve sub-1.2s on warm inference paths.
What is LiveKit and why do you use it for Voice AI?
LiveKit is an open-source WebRTC infrastructure platform with a dedicated Agents SDK for building AI voice bots that connect to real phone calls via SIP. CelloIP uses LiveKit because it handles WebRTC media, SIP bridging, and AI pipeline orchestration in a single framework, reducing integration complexity vs building custom Asterisk AGI pipelines.
Can an AI voice agent handle complex multi-turn conversations?
Yes. CelloIP's voice agents maintain conversation context across turns using LangChain or custom conversation memory, handle clarification questions, perform tool calls (database queries, API lookups, calendar bookings), and escalate to human agents when confidence is low. Production agents handle 8–12 turn conversations reliably.
Mobile SIP Dialer
6 questions
What platforms do you build SIP dialers for?
CelloIP builds SIP dialer applications for iOS (Swift/Objective-C with PJSIP), Android (Kotlin with PJSIP-Android), React Native (cross-platform with PJSIP bindings), and Flutter (cross-platform). All support SIP over TCP/TLS, SRTP encryption, and wideband audio codecs.
Can you customise Linphone or build a white-label dialer app?
Yes. CelloIP customises Linphone (iOS and Android) by modifying the liblinphone SDK and UI layer for white-label branding, custom push notification flows, and enterprise SIP server configs. Alternatively, we build fully custom PJSIP-based dialers from scratch for clients who need complete source ownership.
How do you handle push notifications for incoming VoIP calls on mobile?
For iOS, CelloIP uses Apple PushKit (VoIP push) with CallKit for native dialer UI integration. For Android, FCM (Firebase Cloud Messaging) with a custom SIP proxy push hook. This ensures incoming calls wake the device even when the app is backgrounded, with battery-efficient keepalive mechanisms.
What audio codecs do your mobile dialers support?
CelloIP's mobile dialers support: Opus (wideband, preferred for IP calls), G.711 μ-law/a-law (PSTN interop), G.729 (low-bandwidth), iLBC, and G.722 (HD voice). Codec negotiation is handled automatically based on network conditions and server capability.
Can your mobile dialers work over 3G/4G as well as Wi-Fi?
Yes. CelloIP's dialers are optimised for cellular networks using adaptive jitter buffers, OPUS bitrate adaptation, and aggressive ICE/STUN for NAT traversal. They also handle seamless handoff between Wi-Fi and mobile data using SIP re-registration with minimal audio dropout.
How long does it take to build a custom mobile SIP dialer?
A white-label Linphone customisation takes 4–6 weeks. A fully custom PJSIP-based iOS and Android dialer takes 10–16 weeks. A React Native or Flutter cross-platform dialer takes 8–14 weeks. CelloIP includes push notifications, SRTP, HD audio, and call history as standard features.
Hiring & Staff Augmentation
8 questions
What is the difference between hiring a freelancer and staff augmentation?
Freelancers work independently on defined deliverables with minimal management overhead from you. Staff augmentation means the engineer joins your team, follows your processes, attends your standups, and reports to your tech lead — exactly like an employee but without the permanent commitment or HR overhead. CelloIP provides both models.
How quickly can I onboard a dedicated VoIP developer from CelloIP?
CelloIP can onboard a vetted engineer within 48 hours of contract signing. The process: share your requirements → receive 2-3 matched profiles in 24 hours → optional 1-hour technical interview → NDA and contract signed → developer starts next business day.
Do CelloIP developers sign NDAs?
Yes. NDA is signed on Day 1, before any code or project discussion. CelloIP uses mutual NDAs covering source code, architecture, business logic, client data, and pricing. Individual developer NDAs are also available on request.
Who owns the code written by CelloIP developers?
You do. Full IP assignment is included in every CelloIP contract. All source code, configurations, documentation, and assets created during the engagement are your property from day one, with no licensing restrictions or ongoing royalties.
What engagement models does CelloIP offer?
CelloIP offers three models: (1) Fixed-Price Project — defined scope, milestone-based payments; (2) Dedicated Developer — monthly retainer (160 hrs/month) with a named engineer on your team; (3) Team Augmentation — 2–6 engineers for a defined project duration. Hybrid models (part fixed, part retainer) are available.
Can I scale my team up or down during a project?
Yes. CelloIP's staff augmentation model is designed for elasticity. You can add engineers when workload spikes (e.g., a major feature sprint) and scale back to a maintenance retainer between releases. Minimum commitment is 1 month; notice period for scaling down is 2 weeks.
What time zones do CelloIP developers work in?
CelloIP's team is based in Ahmedabad, India (IST, UTC+5:30). Developers can accommodate US East Coast overlap (mornings IST = late afternoon EST) or European overlap (afternoons IST = morning CET). All engineers are proficient in asynchronous collaboration via Slack, Jira, and GitHub.
How does CelloIP vet its engineers?
CelloIP's vetting process: (1) technical screening (SIP/VoIP theory, live coding in Asterisk/FreeSWITCH/WebRTC); (2) code review of a sample project; (3) communication assessment; (4) background verification. Only engineers with 5+ years of production VoIP experience are placed on client engagements.
Pricing & Engagement Models
6 questions
What are CelloIP's hourly rates for VoIP developers?
CelloIP's hourly rates for dedicated VoIP engineers: Junior (3–5 years) — $25–$32/hr; Mid-level (5–8 years) — $32–$45/hr; Senior Architect (8+ years) — $45–$65/hr. These are 40–60% below equivalent US/UK agency rates. Monthly retainers (160 hrs) start from $4,000/month.
What is included in a monthly retainer engagement?
A monthly retainer includes: 160 hours of dedicated development, daily standups (or async updates), weekly progress reports, code committed to your repository, sprint planning participation, and 5 hours of free bug-fix support post-delivery. NDA and IP assignment are included at no extra cost.
Do you offer fixed-price project contracts?
Yes. For well-defined projects (Asterisk PBX setup, FreeSWITCH softswitch, mobile dialer), CelloIP provides fixed-price contracts with milestone payments (30% kickoff, 40% mid-delivery, 30% on completion). Fixed-price projects include a 30-day post-launch bug-fix warranty.
Is there a minimum contract duration for staff augmentation?
The minimum engagement is 1 month (160 hours). Most clients start with a 3-month pilot and extend to 6–18 months. There is no penalty for early termination after the minimum period, with 2 weeks notice.
What payment methods does CelloIP accept?
CelloIP accepts bank wire transfer (USD, EUR, GBP), PayPal, and Wise for international payments. Invoices are issued monthly for retainers and per-milestone for fixed-price projects. Indian clients can pay via NEFT/RTGS in INR.
Can I get a free proof-of-concept before committing to a full project?
Yes. CelloIP offers a free 2-hour technical consultation to scope your project, followed by an optional paid PoC engagement (typically 40 hours / 1 week) to validate the architecture before committing to a full build. Many clients proceed directly to full development after the consultation.
Security & Compliance
6 questions
Is VoIP communication secure?
VoIP can be made highly secure. CelloIP implements TLS (Transport Layer Security) for SIP signalling and SRTP (Secure Real-Time Transport Protocol) for media encryption on all production deployments. Without these, VoIP is vulnerable to eavesdropping and man-in-the-middle attacks. CelloIP also implements fail2ban, SIP firewall rules, and rate limiting against brute-force attacks.
How do you implement SRTP and TLS for VoIP encryption?
CelloIP enables SIP-TLS on port 5061 (or WSS for WebRTC) using Let's Encrypt or client-provided certificates on Asterisk/FreeSWITCH. SRTP is negotiated via SDES or DTLS-SRTP (mandatory for WebRTC). Media keys are renegotiated per call. Asterisk config: encryption=yes + media_encryption=sdes in PJSIP endpoints.
Can CelloIP build HIPAA-compliant communication solutions?
Yes. CelloIP has built HIPAA-compliant VoIP systems for healthcare clients including: end-to-end SRTP/TLS encryption, BAA (Business Associate Agreement) support, call recording with AES-256 encryption at rest, access controls (role-based), audit logging, and automatic data retention policies. We recommend consulting your compliance officer for final HIPAA certification.
What is STIR/SHAKEN and do your systems support it?
STIR (Secure Telephone Identity Revisited) and SHAKEN (Signature-based Handling of Asserted information using toKENs) are FCC-mandated call attestation standards that reduce robocall spoofing. CelloIP implements STIR/SHAKEN on Asterisk and FreeSWITCH using open-source libraries (libstir-shaken) and commercial attestation services for carriers and MVNOs.
How do you protect against VoIP fraud and toll fraud?
CelloIP implements a multi-layer fraud prevention stack: (1) SIP authentication (digest + TLS); (2) fail2ban for SIP brute-force; (3) rate limiting per source IP via OpenSIPS pike module; (4) international call restrictions and DISA controls; (5) real-time CDR anomaly detection for unusual call patterns; (6) immediate alerting on threshold breaches.
Is call recording compliant with GDPR?
Call recording compliance requires: notifying callers at the start of the call (CelloIP implements auto-announcement), storing recordings encrypted at rest (AES-256), defining a retention period and auto-deletion policy, and providing deletion on request. CelloIP builds GDPR-compliant call recording systems with all these controls as standard.
Project Process & Timeline
6 questions
What does the typical engagement process look like?
CelloIP's process: (1) Free 2-hour technical consultation; (2) Requirements document and architecture proposal (1 week); (3) Fixed-price quote or retainer proposal; (4) NDA + contract signing; (5) Kickoff and environment setup (Week 1); (6) Agile sprints with weekly demos; (7) UAT and staging; (8) Production deployment + handover; (9) 30-day post-launch support.
How do you manage remote development projects?
CelloIP uses Jira (or your preferred PM tool) for sprint tracking, GitHub/GitLab for code with PR-based reviews, Slack for daily communication, and Loom/Zoom for weekly demo calls. All code is committed to your repository with descriptive commit messages and pull request documentation.
What project management tools do you use?
CelloIP works with your existing toolset. We are proficient with: Jira, Linear, Trello, Asana (project management); GitHub, GitLab, Bitbucket (version control); Slack, Microsoft Teams (communication); Confluence, Notion (documentation); Figma (design collaboration). We adapt to your stack, not the other way around.
How long does a typical VoIP project take?
Timelines vary by scope: Basic Asterisk PBX (4–6 weeks), FreeSWITCH softswitch (8–12 weeks), WebRTC platform (10–16 weeks), AI Voice Agent (8–12 weeks), MVNO BSS/OSS (24–36 weeks), Mobile SIP Dialer (8–14 weeks). CelloIP provides detailed project plans with weekly milestones at the proposal stage.
Do you provide post-launch support and maintenance?
Yes. CelloIP offers post-launch support in three tiers: (1) Basic — 30-day bug-fix warranty included in all projects; (2) Standard — monthly retainer (20 hrs/month) for ongoing fixes, updates, and minor features; (3) Premium — dedicated engineer on retainer for continuous development and 4-hour SLA on critical issues.
What happens if I'm not satisfied with the work?
CelloIP has a satisfaction guarantee: if deliverables do not meet the agreed specifications (defined in the requirements doc), CelloIP will fix at no extra charge within the 30-day warranty period. For retainer engagements, you can terminate with 2 weeks notice if quality standards are not met. We have a 98% client retention rate across 200+ projects.
Still have questions?
Our VoIP engineers are ready to answer your specific questions — from architecture reviews to pricing estimates. First consultation is free.