Executive summary
SIP trunking is the method by which a PBX —on-premise or cloud-based— connects to the public telephone network (PSTN) using the SIP protocol over the internet, instead of a dedicated physical circuit. Each trunk is, in essence, a virtual line that carries simultaneous calls to and from the public network using packetized voice.
In Mexico and the United States, SIP trunking has already replaced ISDN PRI trunks in most new deployments. A Mexican business with 20 extensions today typically orders between 8 and 15 SIP channels depending on its traffic profile, paying a monthly fee in Mexican pesos plus a per-minute cost to national and international destinations.
This guide covers:
- What a SIP trunk actually is and how it works technically
- Common codecs and their impact on voice quality
- Pricing, number portability and compliance in the Mexican market
- When it makes sense to migrate from legacy lines
- How to evaluate providers without falling into abusive contracts
Primary CTA: Ready to quote your SIP trunk in Mexico? Get your quote in 24h and receive a personalized proposal covering channels, DIDs and per-minute rates.
What is a SIP trunk and how does it differ from a traditional line?
A SIP trunk is a logical channel between your PBX and a SIP carrier that carries multiple simultaneous calls over an IP connection. The protocol that makes it work is SIP — Session Initiation Protocol, originally defined by the IETF in RFC 3261 [IETF RFC 3261, https://datatracker.ietf.org/doc/html/rfc3261, accessed 2026-09-06].
Conceptually, a SIP trunk replaces what an E1/T1 (PRI) cable or a group of analog lines used to do. Instead of a physical cable with circuit-switched channels, you get an IP session authenticated with your carrier that accepts and originates calls to the public PSTN.
The three pieces of the stack
A production SIP trunk connection has three interacting components:
- SIP signaling: carries the control messages — INVITE to start a call, BYE to terminate it, REGISTER to authenticate. It’s plain text over UDP/TCP/TLS, defined in RFC 3261 [IETF RFC 3261, https://datatracker.ietf.org/doc/html/rfc3261, accessed 2026-09-06].
- RTP (Real-Time Transport Protocol): carries the packetized audio between the two parties. Not defined by SIP itself but by RFC 3550 from the IETF.
- NAT traversal: in networks behind NAT, ICE (Interactive Connectivity Establishment, RFC 5245) negotiates media paths [IETF RFC 5245, https://datatracker.ietf.org/doc/html/rfc5245, accessed 2026-09-06].
Cisco, in its official CUBE documentation, describes it as “a SIP channel that uses standard SIP INVITE and BYE messages to establish and terminate multimedia sessions between endpoints” [Cisco, https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-understanding-sip.html, accessed 2026-09-06]. 3CX puts it more practically: “a SIP trunk is the IP version of an analog phone line, with the difference that it can carry multiple simultaneous calls” [3CX, https://www.3cx.com/learn/what-is-sip-trunk/, accessed 2026-09-06].
How many simultaneous calls fit in a SIP trunk?
A SIP trunk doesn’t have a hard channel limit in the protocol itself: the limit is defined by the commercial agreement with your carrier. In Mexico, typical packages range from 4 channels (micro-business) to 50+ channels (call center). The number of concurrent channels is, in practice, an operational agreement — what matters is the available bandwidth (each G.711 μ-law call consumes roughly 80 kbps including RTP/UDP/IP overhead) and the PBX’s ability to sustain that many concurrent sessions.
Codecs: what affects voice quality
The codec defines how voice gets compressed to travel across the network. The choice directly impacts bandwidth, perceived quality and compatibility with voice agents or STT/TTS pipelines.
| Codec | Bitrate | Typical MOS | When it fits |
|---|---|---|---|
| G.711 μ-law (ulaw) | 64 kbps | ~4.5 | Default MX/US, maximum compatibility, no compression |
| G.711 A-law | 64 kbps | ~4.5 | European variant, same quality |
| G.722 | 64 kbps | ~4.5 | Wideband / HD voice, requires compatible endpoint |
| G.729 | 8 kbps | ~4.0 | Low bandwidth; audible distortion in STT, avoid for voice agents |
| Opus | 6-510 kbps adaptive | ~4.3+ | Voice agents, WebRTC, ideal for STT/TTS in AI pipelines |
For traditional SIP trunk (MX/US): G.711 μ-law is the safe default. Every carrier, every PBX and every IP phone supports it. It costs bandwidth but eliminates surprises.
For voice agents or WebRTC: Opus is the right codec. It scales from 6 kbps (saturated network) up to 510 kbps (HD voice / music), adapts dynamically to the network, and modern STT models (Whisper, Deepgram, OpenAI Realtime) handle it without perceptible distortion.
Practical recommendation: if your business has a traditional PBX with hardware phones, G.711 μ-law. If you’re planning to integrate voice agents or record calls for STT analysis, ask your carrier for explicit Opus support — or at minimum G.722.
SIP trunking market in Mexico (2026)
Typical pricing
The price range in Mexico for monthly SIP channels varies by carrier, contracted volume and whether minutes are included. Market data as of the first half of 2026:
- Monthly SIP channel: MXN $200-400 per channel (typical MX market range, Sep 2026)
- MX geographic DID setup: MXN $250-500 one-time
- MX DID monthly: MXN $80-150/month
- National E.164 minute: MXN $0.05-0.15/min
- International minute: MXN $0.50-1.50/min
These ranges are typical but not universal — premium carriers charge more, retail carriers compete on price and sacrifice support.
Comparison with US market
| Concept | MX carrier | US carrier |
|---|---|---|
| Monthly SIP channel | MXN $200-400 (~$11-22 USD) | $5-15 USD/month |
| DID setup | MXN $250-500 | $5 USD |
| DID monthly | MXN $80-150 | $1.50 USD/month |
| Inbound minute | MXN $0.05-0.15 | $0.005-0.012 USD |
US ranges sourced from [Telnyx pricing, https://telnyx.com/pricing, accessed 2026-09-06] and [Flowroute pricing, https://www.flowroute.com/pricing, accessed 2026-09-06].
The cost difference reflects market size, regulation and bargaining power. A Mexican business with 15 channels pays roughly twice as much in USD as its US counterpart, although the per-minute cost in Mexico is higher in relative terms to local salaries.
Number portability
Mexico has guaranteed number portability since 2011 through the NOM-189-SCT standard. The process takes 5 business days once initiated and applies to both geographic DIDs (52 55 CDMX, 52 33 Guadalajara, 52 81 Monterrey, etc.) and mobile numbers.
Practical implication: if your business changes SIP trunk provider, you keep your numbers. This eliminates the main FUD around migrations (“we’ll lose the number our customers already know”).
Compliance: what IFT and FCC require
Mexico
The Mexican regulatory framework is composed of:
- IFT (Instituto Federal de Telecomunicaciones) — sector regulator
- Ley Federal de Telecomunicaciones y Radiodifusión (LFTR, 2014) — framework legislation
- NOM-189-SCT — number portability
- E911 mandatory since 2019 — location-based routing is the carrier’s responsibility, not the customer’s
United States
If your Mexican business has operations or customers in the US, keep in mind:
- E911/NG911 — the FCC mandates Kari’s Law (2018) and RAY BAUM’s Act (2020), which require location-based routing for emergency calls
- STIR/SHAKEN — attestation mandatory since June 2023 for US carriers [FCC 20-42, https://www.fcc.gov/document/stir-shaken-implementation-deadline-extensions, accessed 2026-09-06]
- TRACED Act (2022) — reinforces the anti-robocalling framework
The operational difference between both markets: in the US, location-based routing uses GIS (geographic information systems) for dispatch; in Mexico, the PSAP (Public Safety Answering Point) receives an approximate location provided by the carrier.
When does migrating to SIP trunking make sense?
There are three clear signals that your business is ready to migrate from legacy lines to SIP trunk:
1. Your monthly phone bill is growing without correlating with usage
Analog lines and PRI have a high fixed cost independent of traffic. SIP trunk lets you buy channels on demand and pay minutes separately. Businesses with marked seasonality (legal, retail with peak seasons) are natural candidates.
2. You have remote employees or multi-site operations
A SIP trunk at a central office can be extended with softphones or IP phones to any location with good connectivity. This eliminates the cost of dedicated lines per branch and unifies numbering.
3. You want to integrate voice agents or call analytics
If you’re planning to automate calls with AI, transcribe calls into your CRM, or run outbound campaigns, you need a SIP trunk over IP — analog lines make that integration impossible.
For a more detailed analysis of migrating from PRI to SIP, including the cost comparison, see SIP trunk vs PRI: when to migrate and what it costs.
How to evaluate providers without falling into traps
Red flags when evaluating carriers
- 24+ month lock-in contracts without a proportional exit clause. Acceptable in enterprise plans with aggressive discount, questionable in SMB plans.
- Bundled minutes that can’t be reviewed at the end of the month. If the carrier doesn’t give you daily CDR (Call Detail Records), you don’t know what you’re paying for.
- Hidden charges for “unlisted” or “premium” DIDs. Negotiate flat rate per standard DID from the start.
- Email-only support. For business telephony, 24/7 support with measurable SLA must be in the contract.
Concrete questions for your short-list
- How many channels do I need based on my peak traffic? (measure with historical CDR)
- What codec do they negotiate by default? (G.711 μ-law is the correct answer for legacy)
- Do they use IP-only auth or digest auth? (IP-only is simpler and more stable)
- What is the measured uptime SLA and the compensation for outages?
- Do they provide daily CDR and in what format?
- How much does it cost to port my existing DIDs?
Checklist before signing a contract
- Daily CDR accessible from carrier portal
- Default codec documented in contract (G.711 μ-law typical)
- Uptime SLA ≥ 99.9% with compensation
- DID portability at no cost or with explicit one-time fee
- 24/7 support with measurable response time
- No forced lock-in, or with proportional exit clause
- IP-only auth or digest auth depending on your PBX
- Compatibility with your PBX (Asterisk, 3CX, FreePBX, Cisco, Avaya, etc.)
Next steps
SIP trunking in Mexico is a mature market in 2026 — carriers compete on price, PBXs support the protocol by default, and number portability eliminates the main barrier to change.
If your business is evaluating a migration, the natural next step is to quote at least two carriers to compare not just channel pricing but also support, measured audio quality and contract flexibility.
Primary CTA: Compare carriers and receive a proposal in 24h. Quote your SIP trunk here and a specialist will reach out with pricing scenarios covering channels, DIDs and per-minute rates tailored to your traffic.