If you've ever shipped a voice feature and watched a call connect with no audio, or sound choppy on a link that passes every speed test, this one is for you. Here's the packet-level view of a VoIP call: what it costs on the wire, where the delay comes from, and which rules apply when the call touches India.
VoIP (Voice over Internet Protocol) sends speech as small digital packets over an IP network instead of a reserved circuit. A VoIP call samples and compresses your voice, sets up the session with a protocol such as SIP, then streams audio over RTP. WhatsApp calls, softphones, and voice AI agents all use it.
This guide shows what happens to those packets, works out what a call really consumes, sets delay against the ITU's limits, helps you choose a model, and reads India's rules from the regulators' own documents.
Note: Key takeaways
- A VoIP call has two planes. Signalling sets up the call and media carries the sound, so a call can connect and still have no audio.
- Headers change the bandwidth math. G.729 compresses voice 8 times more than G.711, yet uses only about 2.8 times less bandwidth on Ethernet.
- Delay and jitter usually matter more than speed. ITU-T G.114 treats 150 ms one way as nearly transparent and 400 ms as the general planning limit.
- VoLTE is VoIP with an operator's guarantees. The packets are similar, but the operator controls the path.
- India's rules turn on the provider's authorisation. Under the June 2026 rules, access services may connect internet telephony to phone networks, while internet service authorisations can't do so within India.
What is a VoIP call?
A VoIP call is any voice session whose audio travels as IP packets for at least part of its route. That definition is broad, and it is why "is this a VoIP call?" often has a surprising answer. The useful questions are who controls the path and whether the call can reach ordinary phone numbers.
| Kind of call | Where IP is used | Who controls the path | Reaches ordinary numbers? |
|---|---|---|---|
| App to app, such as WhatsApp or Meet | End to end | Nobody manages it end to end | Not by itself |
| Softphone or IP desk phone | From your device to the provider | Your network, then the provider | Yes, through a gateway |
| Business call over a SIP trunk | From your system to the provider | Your network, the provider, then the called operator | Yes |
| VoLTE on a 4G phone | Inside the operator's IMS network | The mobile operator | Yes |
| Mobile number over Wi-Fi | Over your Wi-Fi, then the operator's network | Your Wi-Fi, then the operator | Yes |
| Analog landline | Not on your line, though a carrier may use IP inside its network | The carrier | Yes |
Is VoLTE the same as VoIP?
VoLTE is a managed form of VoIP. It also carries digitised voice as IP packets, but it runs on the operator's IP Multimedia Subsystem rather than the open internet. That control lets the operator commit to quality levels, which an app on best-effort internet can't do. The GSMA document IR.92 defines the minimum profile.
What changes when voice moves to packets?
The older phone network reserves a path for the whole call. VoIP gives that up in exchange for sharing, and nearly every strength and weakness of VoIP follows from this one trade.
Sharing means a call can be created, routed, and recorded in software. It also means the network makes no promise about when each packet will land.
How does a VoIP call travel from mouth to ear?
Each call is two separate conversations. Signalling agrees what the call is, and media carries the sound. The diagram follows one direction of audio.
-
Negotiate. SIP, defined in RFC 3261, starts the session. Its
INVITEcarries an SDP description, and the two sides settle on a codec and addresses through the offer/answer model. - Sample. The sender measures the microphone signal thousands of times a second, usually 8,000 for narrowband telephony and 16,000 for wideband.
- Encode. A codec reduces those samples to a target bit rate.
- Packetise. Audio is grouped into frames, commonly 20 ms each, which makes 50 packets a second in each direction. RTP, defined in RFC 3550, adds the sequence number and timestamp the receiver needs.
- Cross the network. Each packet finds its own way past Wi-Fi, routers, and firewalls.
- Buffer and play. A de-jitter buffer holds packets briefly, so sound leaves at an even rate. The decoder then plays them and hides small gaps.
Which codec does each call use?
The codec decides how much of the human voice survives and what the call costs in bandwidth.
| Codec | Audio band | Voice bit rate | Where it fits |
|---|---|---|---|
| G.711 | Narrowband | 64 kbps | The common ground for phone networks, with little processing |
| G.722 | Wideband, 7 kHz | Up to 64 kbps | HD voice between systems that both support it |
| G.729 | Narrowband | 8 kbps | Constrained links where every kilobit counts |
| Opus | Narrowband to fullband | 6 to 510 kbps | Software clients and browsers, and it adapts to the network |
Every WebRTC endpoint must implement Opus and G.711, which is why browser calling can still reach legacy phone systems. Wideband quality rarely survives the whole route, though. A call that leaves IP for the phone network falls to the 8 kHz PSTN floor, so a clear softphone still sounds like a phone at the far end.
Three details that catch engineers
- NAT. A device behind a home or office router may advertise a private address that the far side can't reach. ICE, with STUN and TURN, exists to find a path that works.
- Keypresses. IVR digits usually travel as RFC 4733 telephony events inside RTP, not as tones in the audio. Low-rate codecs can't be relied on to reproduce tones, so both sides need to agree on the method.
- Encryption is two jobs. TLS protects signalling, and SRTP protects media. Enabling one doesn't enable the other.
How much bandwidth does a VoIP call use?
A G.711 call uses about 87 kbps in each direction on Ethernet, not 64. Every 20 ms packet carries 40 bytes of IP, UDP, and RTP headers plus 18 bytes of Ethernet framing. At 50 packets a second, that adds 23.2 kbps no matter which codec you pick.
Our G.711 and G.729 results match Cisco's per-call table. TRAI's February 2025 numbering recommendations use close round figures of 87 and 32 kbps. The Opus bars use the upper end of the bit rates RFC 6716 suggests for speech. Opus varies its rate in practice, so treat them as a planning value.
Four consequences follow:
- Compression pays less than it appears to. G.729 cuts voice to one eighth of G.711, but it cuts the wire rate to about 36%. Headers are 74% of a G.729 stream.
- Longer packets save bandwidth and cost delay. Moving G.729 from 20 ms to 40 ms packets drops it from 31.2 to 19.6 kbps, and adds 20 ms of waiting to each packet.
- SRTP adds a little. Its default 80-bit authentication tag adds about 4 kbps per direction at 50 packets a second.
- Upload often runs out first. A call needs the same rate in both directions, so an asymmetric connection hits its limit on the upload side.
For a worked example, take an office with 40 agents on G.711 with SRTP. That is 40 × 91.2 kbps, or about 3.65 Mbps each way. Leave headroom above that, because voice shares the link with everything else on it.
Why does VoIP sound bad when the speed test looks fine?
A speed test measures how much data moves, while a call cares about when each packet arrives. Three separate measures decide quality: delay, jitter, and packet loss.
ITU-T G.114 gives the reference points for delay. Below 150 ms mouth to ear, most applications feel essentially transparent. Above 400 ms is unacceptable for general network planning, with rare exceptions such as a double satellite hop. For conversational speech, it recommends the E-model to estimate how delay affects quality.
The illustration shows why jitter is expensive. An unsteady network forces a bigger buffer, and that buffer adds delay even when packets aren't late. Trim the buffer too far and late packets get dropped instead, so you trade lag for gaps.
Use the symptom to find the fault:
| What callers report | What to measure | Likely cause | Try first |
|---|---|---|---|
| People talk over each other | One-way delay | A long media route or an oversized jitter buffer | Shorten the media path and review buffer settings |
| Robotic or choppy speech | Jitter and packet loss | Busy Wi-Fi or a saturated uplink | Move to a wired link and prioritise voice |
| Call connects, one way or no audio | RTP packet counts per direction | NAT, a firewall, or a private address in the SDP | Check media firewall rules and the SDP address |
| IVR misses keypresses | DTMF method on each leg | The two sides disagree on telephony events | Align the DTMF setting |
| Fine for hours, then poor at peak | Uplink use against call count | Too little upload for the concurrent calls | Re-run the bandwidth math and add headroom |
Does QoS fix VoIP call quality?
QoS helps on links you control. RFC 4594 recommends the Expedited Forwarding class for telephony, and you can apply it on your LAN and private WAN. Once packets leave your network, RFC 7657 warns that those markings may be changed or removed. Beyond your edge, route choice and provider connectivity matter more than the marking.
Measure the results rather than guess. RTP's companion protocol, RTCP, reports jitter and cumulative loss, and many providers expose them per call.
What changes when an AI agent is on the call?
A voice agent adds speech recognition, model, and speech synthesis time to the same delay budget that telephony draws from. Every millisecond the network uses is one the conversation can't spend on thinking. The agent also needs to hear a caller interrupting while it speaks, which calls for audio flowing both ways at once.
Which type of VoIP should you use?
Most comparison lists sort VoIP by device. A better way is to decide who owns the call logic, because that decides what you run and what you buy.
| Model | You run | The provider runs | Usually billed by | Watch for |
|---|---|---|---|---|
| App to app | The client app | Calling service, or your own relay servers | Often free to users | No reach to ordinary numbers without a gateway |
| SIP trunk | PBX, security, and network | Carrier connectivity | Minutes and channels | Toll fraud on an exposed trunk |
| Hosted cloud PBX | Handsets and softphones | The phone system | Seats | Less control over call logic |
| Voice API | Your application | Telephony and media | Minutes | Your uptime now affects calls |
| Live audio streaming | The speech and model pipeline | Telephony and the media fork | Minutes, plus your AI costs | Turn-taking and interruption are yours to build |
An analog telephone adapter or an IP desk phone is only an endpoint choice within these models. Choose the model first and the device second.
VoIP or landline?
| Analog landline | VoIP | |
|---|---|---|
| Network | Circuit, from your premises | Packets over IP |
| Capacity | Physical lines | Limited by bandwidth and plan |
| Location | Fixed | Depends on the service and its rules |
| Logic and data | Inside the exchange | Available to software through APIs and call records |
| Power cut | Handsets can draw power from the line | Needs power at the router and device |
| Main quality risk | Line faults | Delay, jitter, and packet loss |
Compare costs by line item rather than headline rate. Count per-minute or per-seat charges, numbers, concurrent channels, recording, internet capacity, and the engineering time to run it.
Plan for two gaps early. Put a UPS on the router and switch, and add a second internet path or a mobile fallback. Then settle emergency calling, covered next.
How do emergency calls work on VoIP?
A landline number maps to a physical address, whereas a VoIP device can move, so location needs explicit handling. Rules vary by country. In the US, 47 CFR 9.11 requires interconnected VoIP providers to supply E911 service. India's June 2026 rules say an internet telephony provider may facilitate emergency calls using location services, and must tell users about the limits in unambiguous terms. Don't assume a VoIP line can reach 112 until your provider confirms it.
Is VoIP legal in India?
Yes, but what's allowed depends on which authorisation the provider holds and how the call is carried. The rules changed in June 2026, so older guides may be out of date. Go to the regulators' documents:
| Question | What the source says | Source and date |
|---|---|---|
| What framework applies now? | Five principal authorisations: unified, access, wireline access, internet, and long distance | DoT Gazette notification G.S.R. 513(E), rule 4, 23 June 2026 |
| Who can offer full internet telephony? | The access authorisation includes it, for network operators and virtual network operators. So does wireline access, which only virtual network operators can hold, and the unified authorisation | G.S.R. 513(E), rules 65, 69, and 70 |
| Can it connect to phone networks? | A network operator may interconnect it with PSTN and mobile networks. A virtual network operator connects only to its parent operator, which handles interconnection | G.S.R. 513(E), rule 71(4)(i) and 71(5)(b) |
| Which devices and numbers? | Fixed or mobile terminals. A provider may give one user the same identifier for mobile and mobile internet telephony, or for fixed and fixed internet telephony | G.S.R. 513(E), rule 71(3) |
| What can an internet service provider offer? | A limited service: computer to computer, a device in India to phones outside India, and static-IP devices to similar devices. It can't connect users to or from phone networks in India, or use E.164 numbers | G.S.R. 513(E), rules 74 and 75(1)(b) |
| What about existing licences? | Holders aren't forced into immediate replacement, and they may migrate to the corresponding authorisation | DoT eServices portal, July 2026 |
| What did the older licences allow? | Basic, UASL, CMTS, and Unified Licence (access) holders could offer it unrestricted and interconnect it with PSTN and mobile networks | TRAI response to DoT, July 2018 |
| Did it have to run on the operator's own internet access? | Not under those licences. DoT clarified that the service is untethered from the access network | DoT clarification quoted by TRAI, 19 June 2018 |
| What can a call centre not do? | For OSPs, carrying public network calls between Indian cities, or between India and abroad, over their own network is illegal toll bypass | DoT OSP guidelines, June 2021 |
The 2026 rules don't mention OSPs, and we found no later change to the 2021 guidelines. Separate rules or orders may still apply, so check with counsel before you design a route.
In practice, ask any Indian provider four questions:
- Which authorisation covers this service? Ask for the specific category and the migration status from the old licence.
- How does the call reach the phone network? Ask where it leaves IP, and through whose interconnect.
- Do both legs stay in India? Ask how the provider treats this. Vobiz, for example, applies a media anchoring rule to Indian calls.
- Which number series fits this traffic? Commercial calling has its own rules, including series for promotional and service calls. Our SIP trunking guide covers them with sources.
This section is general information, not legal advice.
How Vobiz delivers VoIP
Vobiz covers the SIP trunk, Voice API, and live audio models above on one account, so you can choose by how your application needs to handle calls.
-
SIP trunks. Outbound and inbound trunks connect a PBX or voice platform. The
transportfield controls signalling, including TLS, and thesecurefield controls SRTP for media. - Your own carrier. With bring your own carrier, you connect an existing carrier relationship. That page lists G.711, G.729, and Opus support.
- Programmable calls. The Voice API and VobizXML drive call flow through webhooks (callbacks), which your application receives and answers.
- Live audio. Audio Streams fork a call to your WebSocket server. Inbound audio is L16 at 8 or 16 kHz, or μ-law at 8 kHz, and bidirectional mode accepts L16 at up to 24 kHz in return.
- Voice AI platforms. Trunks connect to platforms such as LiveKit and Vapi. SIP vs WebSockets helps you choose the path.
- Browsers and soft-phones. Registered clients use endpoints, and the WebRTC Playground places browser calls.
-
Quality data. Each call detail record reports
mos,jitter,packet_loss, and the negotiatedcodec, so you can compare this guide's targets with real calls. - Firewalls. The IP whitelisting page lists current signalling and media ranges.
- India. Calling regulations cover KYC, caller ID, number series, and media anchoring.
- Reach and rates. Current rates are on the Vobiz pricing page.
Test calls are billed and reach real people. Run them with your own Auth ID and Auth Token, and confirm numbers first.
Frequently asked questions
FAQ
What is the full form of VoIP?
VoIP is short for Voice over Internet Protocol. The term means carrying voice as data packets over an IP network, which can be the public internet or a private network, rather than over a reserved telephone circuit.
Is a WhatsApp call a VoIP call?
Yes. A WhatsApp call between two app users sends audio as IP packets over your data or Wi-Fi connection, so it is VoIP. It stays inside that app, unlike a business VoIP service that connects through a gateway to ordinary phone numbers.
Is VoLTE the same as VoIP?
VoLTE is a managed type of VoIP. Both carry voice as IP packets, but VoLTE runs on the mobile operator's IMS network, where the operator can commit to quality levels. An app on the open internet gets best-effort delivery instead.
How much internet speed does a VoIP call need?
Plan about 87 kbps each way for a G.711 call on Ethernet, or about 31 kbps for G.729. Multiply by your concurrent calls and leave headroom. Steady delay and low packet loss matter more than a high speed test result.
Is VoIP legal in India?
Yes, though what's allowed depends on the provider's authorisation and how the call is carried. DoT's June 2026 rules let access providers connect internet telephony to phone networks, but limit internet service providers. Toll bypass remains a concern for call centres. Confirm your design with your provider and counsel.
Does VoIP work during a power cut or without internet?
Not on its own. Your router and device need power, and the call needs an IP connection. A UPS keeps equipment running, and a second internet link or mobile fallback covers an outage. Some services can forward calls to a mobile number.
Further reading on Vobiz
- What is SIP? · SIP vs VoIP · What is SIP trunking? · What is WebRTC?
- What is a Voice API? · SIP vs WebSockets · Audio Streams API
- India calling regulations · UCC management · Hangup causes
Sources
Standards
- IETF, RFC 3261: SIP: Session Initiation Protocol, June 2002.
- IETF, RFC 3264: An Offer/Answer Model with SDP, June 2002.
- IETF, RFC 3550: RTP: A Transport Protocol for Real-Time Applications, July 2003.
- IETF, RFC 3711: The Secure Real-time Transport Protocol, March 2004.
- IETF, RFC 4594: Configuration Guidelines for DiffServ Service Classes, August 2006.
- IETF, RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals, December 2006.
- IETF, RFC 6716: Definition of the Opus Audio Codec, September 2012.
- IETF, RFC 7657: Differentiated Services and Real-Time Communication, November 2015.
- IETF, RFC 7874: WebRTC Audio Codec and Processing Requirements, May 2016.
- IETF, RFC 8445: Interactive Connectivity Establishment, July 2018.
- IETF, RFC 8866: SDP: Session Description Protocol, January 2021.
- ITU-T, G.114: One-way transmission time, May 2003.
- ITU-T, G.711: Pulse code modulation of voice frequencies, November 1988.
- ITU-T, G.722: 7 kHz audio-coding within 64 kbit/s, September 2012.
- ITU-T, G.729: Coding of speech at 8 kbit/s, June 2012.
Regulators and law
- TRAI, Response on the regulatory framework for Internet Telephony, 16 July 2018.
- TRAI, Recommendations on Revision of National Numbering Plan, 6 February 2025.
- DoT, Telecommunications (Authorisation for Provision of Principal Telecommunication Services) Rules, 2026, G.S.R. 513(E), Gazette of India, 23 June 2026.
- DoT, Principal Telecom Services Rules, 2026, eServices portal article, July 2026.
- DoT, Revised Guidelines for Other Service Providers, 23 June 2021.
- Khaitan & Co, The age of internet telephony has officially arrived, 2018.
- US Code of Federal Regulations, 47 CFR 9.11: E911 service, via Cornell LII.
Reference
- Cisco, Modify bandwidth consumption calculation for voice calls.
- Cisco, Understanding delay in packet voice networks.
- EE World Online, VoLTE: what makes voice over IP carrier grade, September 2012.
- Wikipedia, "Voice over IP".
Telephony for voice AI agents: Vobiz — SIP trunks and phone numbers for voice AI.
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