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Connecting Tieline to other Codecs Using SIP
© Tieline Pty. Ltd. 2021
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Connecting Tieline to other Codecs Using SIP
To dial between Tieline and non-Tieline codecs over IP it is necessary to configure all codecs to
connect in SIP mode. SIP provides interoperability between different brands of codecs due to its
standardized protocols for connecting different devices. Tieline IP codecs are EBU N/ACIP Tech
3326 compliant when connecting using SIP (Session Initiation Protocol) to other brands of IP
codecs.
SIP is also a useful way of dialing another device and locating it easily. This task is usually
performed by SIP servers, which communicate between SIP-compliant devices to set up a call. SIP
connections can be made in two ways; registered or unregistered.
Unregistered Peer-to-Peer SIP Connections
Codecs don’t need to be registered to a SIP server to dial peer-to-peer SIP connections. An
unregistered SIP peer-to-peer connection involves two codecs connecting to each other directly
using an IP address, as you would for a standard Tieline IP call. This is simpler and much like the
way codecs normally connect. The difference is that a Tieline IP call uses proprietary Tieline
session data to negotiate call parameters (e.g. algorithm and bit rate) when a call is established,
whereas a peer-to-peer SIP connection uses Session Description Protocol (SDP) for this purpose.
SIP provides interoperability between different brands of codecs due to its standardized protocols for
connecting dissimilar devices and is used when connecting Tieline codecs to non-Tieline devices.
There are two very distinct parts to a call when dialing over IP. The initial stage is the call setup
stage and this is what SIP and SDP is used for. The second stage is when data transfer occurs and
this is left to the other protocols such as RTP/UDP to stream audio data. SDP works with a number
of other protocols, to deliver the following functions when connecting devices over SIP:
·
Establish a codec’s location.
·
Determine the availability of a codec.
·
Negotiate the features to be used during a call, e.g. the algorithm and bit rate.
·
Provide call management of participants.
·
Adjust session management features while a call is in progress (e.g. termination and
transfer of calls).
All the mandatory EBU N/ACIP 3326 algorithms are supported in the codec, including G.711,
G.722, MPEG-1 Layer 2 and 16 bit PCM, as well as optional algorithms including Opus, LC-AAC,
AAC-LD, HE-AACv2 and aptX Enhanced.
Registered SIP Server Connections
The benefit of using a SIP server to connect is that any device can be ‘discovered’ via its SIP server
registration. This is particularly useful if a codec is being used in multiple locations with IP
addresses that are DHCP assigned. These DHCP addresses are unreliable and are not
recommended for live broadcast connections. As long as your codec and the device you are dialing
are both registered to a SIP server you can connect by simply dialing the destination SIP address.
