Universal Telecom
Contact

Glossary · VoIP and call quality

SIP

In one sentence

SIP (Session Initiation Protocol) is the standard signalling protocol that sets up, changes and ends VoIP calls. It handles ringing, answering and hanging up; the voice itself travels separately.

Also known as : Session Initiation Protocol · SIP protocol

What is SIP?

SIP, the Session Initiation Protocol, is the common language of internet telephony. Published by the IETF as RFC 3261 in 2002, it opens, modifies and closes communication “sessions” between two or more parties. In business telephony those sessions are mostly phone calls, although SIP can also start video calls or instant messaging.

SIP is why an IP desk phone from one manufacturer, a softphone from another and a PBX from a third can talk to each other and to any carrier. Its strength is interoperability; its weakness is flexibility, which lets vendors interpret some details differently and is behind many “it works with everything except…” support tickets.

How it works

SIP looks a lot like HTTP: plain-text requests and numbered responses.

  1. REGISTER: a handset tells the server “I’m extension 204, reachable at this IP address”.
  2. INVITE: one endpoint asks to call another. The message carries an SDP description of the codecs it accepts and where to send the audio.
  3. Responses follow: 100 Trying, 180 Ringing, then 200 OK when the other side answers. The caller confirms with ACK.
  4. BYE ends the call.

The voice never travels inside SIP: it flows as RTP according to what was negotiated. That is why a firewall problem can produce a call that rings perfectly but has no audio in one direction.

Keypad presses, which an IVR needs, are sent as DTMF, either within the RTP stream (RFC 4733) or as SIP INFO messages. When SIP crosses between networks, a session border controller often sits at the edge to secure and tidy it.

Example

A Cork-based logistics company rolls out IP phones in its three depots. At the Limerick depot every outbound call drops after about 30 seconds. A SIP trace shows that the final ACK never reaches the provider because the depot router’s SIP ALG is rewriting the messages. With the feature switched off, calls stay up. Knowing how to read a SIP exchange turns a vague complaint into a five-minute fix.

SIP at Universal Telecom

Our handsets, virtual PBX and SIP trunk all use standard SIP, so most IP phones and PBX platforms connect without fuss. Carriers and integrators interconnecting their own platforms should look at our wholesale offer.

Frequently asked questions

What ports does SIP use?

By convention port 5060 over UDP or TCP, and 5061 for SIP encrypted with TLS. The audio travels separately as RTP over a range of UDP ports set by the device or provider.

What is SIP ALG and should I turn it off?

SIP ALG is a router feature that rewrites SIP messages to get them through NAT. It often gets it wrong, causing one-way audio, dropped calls and phones that keep unregistering, so most VoIP providers advise disabling it.

Is SIP the same as VoIP?

No. VoIP is the overall approach of carrying voice over IP networks. SIP is the protocol most business VoIP uses to set up and control calls; the voice is carried by RTP.

Related terms

  • VoIP

    VoIP (voice over internet protocol) carries phone calls as packets of data over an IP network, such as the internet or an office LAN, instead of over the copper lines of the traditional phone network.

  • SIP trunk

    A SIP trunk is a voice-over-IP connection between a business phone system and a carrier. It replaces ISDN lines and carries calls and numbers over your internet connection.

  • SBC (session border controller)

    A session border controller (SBC) sits at the edge of a network and controls SIP calls going in and out: it filters traffic, hides the internal network, handles NAT and can transcode or encrypt calls.

  • Codec

    A codec is the algorithm that digitises and compresses your voice during a VoIP call and turns it back into sound at the other end. It sets both the audio quality and the bandwidth used.

  • Softphone

    A softphone is a phone application that runs on a computer or smartphone. It replaces the desk phone and connects to the company phone system over VoIP, using the business number.

  • DTMF

    DTMF (dual-tone multi-frequency) is the signal a phone sends when you press a key during a call: two tones played together for each key, which an IVR or remote system decodes.

Updated on

← Back to the glossary