Tieline Bridge-IT XTRA, Руководство пользователя

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94 Bridge-IT XTRA Manual
© Tieline Pty. Ltd. 2015
10.Click Finish and the newly created program will be displayed in the left pane within the
Programs panel and in the Master panel .
11.Configure multicast server and multicast client programs and load all codecs with the appropriate
program. Select and connect audio streams in a program using the Master panel , or dial the
program manually using the codec front panel. Dial the multicast server program connection first
and then connect multicast client codec programs to begin receiving multicast audio packets.
16.7 Configuring SIP Settings
The codec is fully EBU N/ACIP Tech 3326 compliant when connecting using SIP (Session Initiation
Protocol) to other brands of IP codecs.
About SIP
SIP provides superior interoperability between different brands of codecs due to its standardized
protocols for connecting devices and is intended to be used when connecting Tieline codecs to non-
Tieline devices. Devices primarily use SIP to dial another device’s SIP address and find its location
with a minimum of fuss. This task is usually performed by SIP servers, which communicate between
SIP-compliant devices to set up a call.
When connecting two devices, SDP performs similar tasks to Tieline’s proprietary session data,
which is used to configure all non-SIP IP connections. 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 is used for. The
second stage is when data transference occurs and this is left to the other protocols used by a
device (i.e. using UDP to send audio data).
All the mandatory EBU N/ACIP 3326 algorithms are supported (G.711, G.722, MPEG-1 Layer 2 and
16 bit PCM), as well as optional algorithms including LC- AAC, HE-AAC and aptX Enhanced. The
default algorithm selected when connecting using SIP is G.711.
Important Notes:
· Each codec should be registered to a different SIP server account to avoid connection
conflicts.
· SIP dialing is only supported over peer-to-peer connections, not multi-unicast
connections.
· Tieline G3 codecs do not support AAC and will default to MPEG Layer 2 if a Bridge-IT
codec configured for AAC attempts to connect.
·
Some Telcos and ISPs may block SIP traffic over UDP port 5060.
·
SIP account registration can only be configured via the Ethernet port and not a VLAN.
·
When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level in the codec. To do this select
SETTINGS > Audio > Ref
Level > Tieline G3 . In addition, select the following on the G3 codec prior to dialing.

Select either a mono or stereo profile

Select [Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP]

Select [Menu] > [Configuration] > [IP1 Setup] > [Algorithm] > [G711/G722 or
MP2]
SIP Server Connections: Getting Started
Registering codecs for SIP connectivity is simple. First, choose the SIP server that you wish to
register your codec with. On a LAN this may be your own server, or it could be one of the many
internet servers available. We recommend that you use your own SIP server and configure it to use
G.711, G.722, MP2 and AAC algorithms. This is because most internet SIP servers are for VoIP