Business VoIP telephony: SIP, SIP trunks, softphones and the end of copper in France

In traditional telephony, the public switched telephone network (PSTN, or RTC in French) set up a dedicated circuit between two handsets for the whole duration of the call. In VoIP, the voice is digitised, split into packets and carried over an IP network, either the internet or an operator’s network. “VoIP” and “IP telephony” refer to the same idea; the term ToIP is also used for business telephony based on this principle.

A VoIP call involves two types of flow, as the ANSSI, France’s national cybersecurity agency, points out: signalling flows, which manage exchanges between components and in particular allow calls to be set up, and media flows, which carry the voice, generally using the RTP protocol. This distinction explains most of the settings, and faults, you will come across.

SIP (Session Initiation Protocol) is a signalling protocol very widely used in IP telephony. Its specification, RFC 3261, published by the IETF in June 2002, describes it as an application-layer control protocol for creating, modifying and terminating sessions with one or more participants, including internet telephone calls. It is what makes a handset ring, passes on the caller’s number or triggers a transfer.

A SIP trunk is a connection supplied by an operator between your phone system, whether an on-premises business phone system or a hosted platform, and the public telephone network, allowing you to make and receive calls with your numbers. It replaces old physical lines and is sized by the number of simultaneous calls rather than the number of sockets. The choice between a phone system installed on your premises and a hosted one is covered in our IP PBX vs Centrex comparison.

Finally, a softphone is phone software installed on a computer or smartphone: it replaces the physical handset, with a headset or the phone’s earphones. It lets you answer business calls wherever you are, provided you have a good-quality connection. Its security does require attention, however: in its 2013 technical note, the ANSSI advises against using softphones connected to the telephony infrastructure, which expose it more (loss of confidentiality, intrusion), and asks, if they are used nonetheless, for the networks of the devices using them to be segregated.

Orange, which owns the copper network, has begun closing it progressively under a plan made public in 2022 and monitored by the Arcep, the French telecoms regulator. According to the regulator, the process runs until the end of 2030 in seven annual batches. A large-scale commercial closure, which prevents new subscriptions on copper, took place on 31 January 2026 in nearly three quarters of municipalities, representing nearly half of the population and around 20 million premises according to the Arcep. Orange postponed by one year, to 31 January 2027, the commercial closure of some of the municipalities initially planned, without ruling out further postponements. Technical closures, which cut off the subscriptions still active, then take place in batches: the one for batch 2 took place on 27 January 2026 and the following ones begin in January 2027; in total, technical closure is due to affect more than 40 million premises by the end of 2030.

For landline telephony, the Arcep is explicit: the PSTN, whether analogue or digital, will no longer be available, and the alternatives are voice over IP and collaboration solutions. The regulator encourages businesses to take stock of analogue or digital lines and phone systems running via the PSTN, but also lifts, remote monitoring, payment terminals, emergency lines and fax machines. It points out that some equipment was powered electrically by the copper line, which fibre cannot do, and that migration can take several weeks, or even several months for the whole process. The French Ministry of the Economy’s website and Orange’s interactive map give the deadline for each municipality.

The quality of a VoIP call depends on the network that carries it. Four concepts are enough: the available bandwidth, which must cover all simultaneous calls in addition to other uses; latency, the time it takes the voice to travel, which makes people talk over each other when it is too high; jitter, the variation in that delay from one packet to the next, which makes the voice choppy; and packet loss, which makes fragments of sentences disappear.

The remedies are well known: a good-quality fixed connection, prioritising voice traffic on the router (quality of service, or QoS), a well-sized and secure Wi-Fi network if softphones use it (the ANSSI advises against wireless IP phones and, if Wi-Fi is used, refers to its dedicated recommendations) and up-to-date equipment. The ANSSI also notes that encrypting communications uses more bandwidth and resources, which must be planned for from the sizing stage. The most reliable approach is still to test in real conditions, at busy times, before switching over.

In its technical note on securing IP telephony, dated 23 December 2013, the ANSSI observes that attackers’ main motivation is to misuse telephone resources, in particular to make calls at the compromised organisation’s expense. It recommends regular monitoring of billing, watching for premium-rate calls, international calls, calls made outside working hours and long calls.

The agency advises disabling risky functions by default, then granting them only to the users who need them, and paying particular attention to international calls and call transfer. It recommends protecting signalling and voice with secure protocols, such as SIP-TLS and SRTP. In practice: strong, unique passwords on SIP accounts, regular updates, and international destinations blocked if they are not needed. Hacking of a business’s landline telephony is among the threats covered by the French government’s Cybermalveillance.gouv.fr scheme.

With IP telephony, routing calls to an external service becomes a matter of configuration: forwarding from the operator’s interface, time-based rules, or connection via a SIP trunk for more complete installations. An automated reception, an answering service or an AI agent can thus take calls without any change of number or equipment. Keep the ANSSI’s recommendations on call transfer in mind and always provide a fallback route to a person.

Alyas fits into this setup: the principle is to forward your existing line to the agent, depending on your operator’s call forwarding options, with the phone setup checked before going live. The agent answers 24 hours a day, announces that it is an AI and that the call is recorded, answers from the information you provide, collects contact details, books appointments if Google Calendar or Cal.com is connected, and transfers the call to your transfer number when the caller asks. Call recordings, transcripts and summaries are hosted in the European Union. The trial lasts 10 days, with no credit card required.