Explained: SIP Registration, Peering and IAX2

Tel2 supports SIP registration, SIP peering and IAX2 registration to connect your VoIP service.

SIP registration is the process by which a SIP client (such as a VoIP phone or softphone) notifies a SIP server (such as Tel2 Cloud PBX) about its current location and availability to receive calls. It is commonly used in scenarios where a user has a dynamic IP address or does not have the ability to open firewall ports that SIP Peering requires.

When a SIP client wants to make or receive calls, it registers its current IP address, username, and password with a SIP registrar server. This registration process allows the SIP server to keep track of the client's location and forward calls to the correct destination. 

SIP Registration is the default configuration with Tel2 services. By configuring your IP-phone credentials (found in your CloudPBX portal) and the SIP Server / proxy host details (phone.tel2.co.uk) you’ll be ready to start making calls.

SIP peering, also known as SIP trunking or interconnection, involves the direct connection and communication between two SIP-enabled devices or networks, allowing them to exchange voice and multimedia sessions without the need for intermediaries. 

Peering agreements are established between the involved parties to allow the exchange of traffic. Since this involves the opening of firewall ports to the other party, both networks need to have a trusted relationship.

The main advantage of Peering is greater flexibility of number routing within your onsite-PBX. To use Peering you will require a static WAN IP address.

To use SIP Peering with Tel2 services, please see our configuration guide here.

IAX2 is an Asterisk specific protocol which is an alternative to SIP. IAX2 registration simplifies some aspects compared to SIP registration. 

Single Port Usage:

IAX2 typically uses a single UDP port (default is 4569) for both signaling and voice traffic. This can simplify firewall configuration compared to SIP, which often requires the management of multiple ports.

Note: SIP using TLS and SRTP also accomplished these same benefits.

Efficient Use of Bandwidth:

IAX2 encapsulates both signaling and voice data within a single packet, leading to more efficient bandwidth usage compared to SIP, which may require separate channels for signaling and media. For example, eight G.729 calls using SIP will use around 250kbps but less than 100kbps using IAX2 trunking.

Trunking and Multiplexing:

IAX2 allows trunking multiple voice channels over a single connection, which simplifies management and can be more efficient than SIP, especially in scenarios with high call volume.

NAT Traversal:

IAX2 can be more NAT-friendly than SIP, as it carries both signaling and media over a single port, potentially reducing issues related to Network Address Translation.

Simplified DMTF:

DTMF traffic is always out of band removing any confusion about which DTMF method to use

Limited Adoption:

SIP is a more widely adopted and standardized protocol in the VoIP industry. IAX2 is specific to the Asterisk PBX system, and its usage is not as prevalent in non-Asterisk environments.

Interoperability:

SIP has better interoperability with a wide range of VoIP equipment and software from different vendors. IAX2 may face compatibility challenges when integrating with non-Asterisk systems.

Community and Support:

The SIP protocol has a larger community and more extensive support, which can be beneficial when seeking help, troubleshooting, or finding documentation.