Codec Tieline TLR300 User manual

Tieline G5 Codec Compatibility
over IP and ISDN
Manual Version: 1.2
May, 2019

Tieline G5 Codec Compatibility over IP and ISDN v1.22
© Tieline Pty. Ltd. 2019
Table of Contents
Part I Connecting Tieline IP to other
Codecs Using SIP 3
................................................................................................................................... 81 Connecting to a Comrex AccessRack Codec
................................................................................................................................... 92 Connecting to a Comrex AccessPortable
................................................................................................................................... 103 Connecting to a Mayah Sporty
................................................................................................................................... 114 Connecting to a Telos Zephyr IP
................................................................................................................................... 125 Connecting to an APT Worldcast Equinox
................................................................................................................................... 126 Connecting to an ProdysProntonet LC
Part II Connecting Tieline ISDN to other
Codecs 14
................................................................................................................................... 141 Connecting to APT Wordcast Equinox ISDN
................................................................................................................................... 162 Connecting to CDQ Prima ISDN
................................................................................................................................... 193 Connecting to Mayah ISDN
................................................................................................................................... 214 Connecting to Telos Zephyr Xstream ISDN
................................................................................................................................... 245 Connecting to Comrex Matrix ISDN

3Connecting Tieline IP to other Codecs Using SIP
© Tieline Pty. Ltd. 2019
1 Connecting Tieline IP 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 aptXEnhanced.
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.

4 Tieline G5 Codec Compatibility over IP and ISDN v1.2
© Tieline Pty. Ltd. 2019
Some SIP servers route RTP audio through the SIP server as well and Tieline recommends avoiding
this type of server whenever possible. Otherwise you will be reliant on the SIP server for streaming
broadcast audio packets and most servers are not designed for mission critical packet streaming.
To dial a codec via a SIP server requires:
1. Both devices to be registered with separate SIP accounts.
2. Both codecs configured to operate in SIP mode.
3. The IP address of the SIP server.
4. An IT administrator to open UDP port 5060 to enable SIP traffic, as well as UDP audio port
5004.
A SIP server administrator should be able to provide the following details to enable SIP registration
of a device:
·Username
·Authorized User
·SIP address
·Domain
·Realm
·Registrar
·Registar port
·Outbound Proxy
·Proxy port
Advantages and Disadvantages of Using SIP
Advantages of SIP
1. SIP provides interoperability between different brands of codecs due to its standardized
protocols for connecting different devices, e.g. Session Description Protocol (SDP).
2. EBU N/ACIP Tech 3326 provides a minimum set of requirements necessary to ensure
interoperability between equipment intended for the transport of audio over IP networks. It
standardizes the use of ports, encoders, transport protocols and signaling to ensure codecs,
and other SIP-enabled devices like smartphones and VoIP phones, can connect
successfully.
Disadvantages of SIP
1. SIP requires both codecs to be configured with the same settings before connecting or the
connection will fail. When dialing Tieline to Tieline the dialing codec provides all connection
information to the answering codec. This is not possible when using SDP in SIP mode.
2. Not all professional codec manufacturers are fully compliant with all requirements for
interoperability.

5Connecting Tieline IP to other Codecs Using SIP
© Tieline Pty. Ltd. 2019
3. SIP connections are more complex to configure.
4. SIP does not support advanced software enhancements which deliver redundancy and rock
solid reliablity over IP, e.g. failover connections, SmartStream PLUS redundant streaming,
Fuse-IP bonding, plus error concealment strategies.
5. Codecs using SIP cannot use the TieLink Traversal Server for presence and connections. In
addition port forwarding is usually required.
6. Some ISPs and/or cellular networks may block SIP traffic.
SIP Security
Tools such as Shodan make it easy for anyone to easily locate devices connected to the internet
around the world. Therefore it is critical that security measures are in place for all IP and SIP
connections over the public internet.
Managing Unwanted SIP Calls
Hackers and other nefarious net-bound characters look for networks with easy access in which to
ply their trade. As a starting point they look for networks with open gateways and platforms using
default passwords.
Maintaining Codec Network Security
Adequate security is a major factor in ensuring your codecs and your broadcast network remain
secure. There are several layers of security available in Tieline codecs to maintain secure
connections. These include:
1. Immediately change the default password when you commission and install your codecs
(see instructions which follow). Create a strong password which includes both capital and
lower case letters, symbols and numbers (up to 15 characters can be entered). Password
managers can be useful when managing multiple passwords within organizations.
2. Ensure your codec is behind a firewall and only open the TCP and UDP ports required to
transmit session and audio data between your codecs. Using non-standard ports instead of
Tieline default ports can also ensure the codec is more difficult to discover. SIP usually uses
standard ports; UDP Port 5060 for session data (SDP) and UDP Port 5004 for audio data.
3. Ports 80 and 8080 are commonly used to access the Tieline codec web server. You can add
an additional layer of security by translating these ports on the WAN side of your network
into non-standard port numbers. Adjust ports using the Options panel in the Toolbox
HTML5 Web-GUI, or see "Configuring TCP/UDP Ports".
4. By default SIP interfaces are disabled to avoid unwanted traffic. The SIP Filter Lists panel
in the Toolbox HTML5 Web-GUI allows filtering of SIP URIs and User Agents to provide
greater security when using SIP. See "Configure SIP White and Blacklists" for more
information.
5. An SSL security certificate can be installed on each codec in your network to ensure it is a
trusted device within your network. See "Installing a Security Certificate" for more
information.
6. Firewall settings to enable or disable a range of firewall-related network services, or limit ping
to only work in a local subnet. Adjust settings using the Options panel in the Toolbox
HTML5 Web-GUI, or see "Firewall Configuration."
7. Implementation of CSRF protection (Cross-Site Request Forgery). Enable and disable this
setting using the Options panel in the Toolbox HTML5 Web-GU, or see "Enabling CSRF
Security" for more info.
Be sure to document any port changes because this information will be required if you need to
contact Tieline or other online support services.

6 Tieline G5 Codec Compatibility over IP and ISDN v1.2
© Tieline Pty. Ltd. 2019
Understanding SIP Terminology
What is a URI?
A Uniform Resource Identifier (URI) is a string of characters identifying a name or resource on the
internet. A Uniform Resource Locator (URL) is a URI specifying where an identified resource is
available (e.g. network location) and the mechanism for retrieving it. E.g. http://twitter.com, where
http:// is the protocol used to retrieve information from "twitter.com" which is the network location.
What is a SIP URI?
SIP URIs can be used to whitelist or blacklist devices from connecting to a codec. A SIP URI is the
address or characters used to call another person via SIP. A SIP URI is essentially a user's 'contact
address' and is used by VoIP phones and other devices to call using SIP. An example of a SIP URI
is: "sip:[e-mail]". When dialing using Tieline codecs "sip:" is automatically
placed in the dial string and does not need to be entered. SIP URIs can be used to whitelist or
blacklist devices from connecting to a codec.
Important Notes:
·The default UDP audio port when using SIP for a peer-to-peer connection is 5004 in
Tieline codecs.
·Enter the IP address or SIP URI, then a full colon and the session port number to
change the session port from the default setting of UDP port 5060, e.g.
"sip:[e-mail]:5070"
·Tieline codecs automatically add "sip:" to the address entered in the Address field
when dialing, so it's not necessary to add this.
·To only allow a predefined list of codecs to connect, add them to the URI Whitelist and
add a wildcard (asterisk) * to the URI Blacklist: all incoming calls will be blocked
except for codecs in the Whitelist.
What is a SIP User Agent?
A user agent is a software agent acting on behalf of a user. Hackers use SIP user agents or
'botnets' to scan for open ports on the internet. When they locate open ports they can force their
way into SIP servers and scan for valid accounts to use for fraudulent purposes, like making free
international calls. Known user agents like "sipvicious" and "friendly-scanner" can be added to a
User Agent Blacklist in Tieline codecs to stop them from accessing them.

7Connecting Tieline IP to other Codecs Using SIP
© Tieline Pty. Ltd. 2019
How do I find myCodec's User Agent
It may be necessary to discover a codec user agent as some codec manufacturers allow
whitelisting of calls by 'User Agent.' Therefore, it may be necessary to enter a Tieline codec
user agent into a non-Tieline codec to allow it to connect to a Tieline SIP codec.
·From firmware v2.16.xx and later releases the user agent in a Tieline G5 codec is configured
as "Tieline <ProductName> <Firmware Version>". E.g. Tieline Bridge-IT v2.18.68.
·In Tieline G3 codecs the user agent is configured as "Tieline <ProductName> <Serial
Number>". E.g. Tieline TLR350 8972. The model numbers for Tieline G3 codecs are as
follows:
§Commander G3 Rack TLR300 = Model Number TLR300
§Commander G3 Rack TLR300B = Model Number TLR350
§Commander G3 Field TLF300 = Model Number TLF300
§i-Mix G3 TLM600 = Model Number TLM600
Troubleshooting SIP Connections
Manufacturers of professional IP codecs in most cases do not use the standard SIP ports (UDP
5060 and UDP 5004) for making IP connections. Therefore you will need to reconfigure each codec
to use the standard SIP ports in most instances.
Most of the time EBU N/ACIP Tech 3326 SIP compliant codecs should connect when using the
same encoders and connection settings at both ends. However, if one-way audio is encountered, it
is highly likely to be a port forwarding issue. Port forwarding configuration instructions for most
popular routers are available at www.portforward.com.
Some manufacturers support a subset of the EBU N/ACIP Tech 3326 recommendations so not all
algorithms are supported by all manufacturers. G.722 is supported by most codec manufacturers,
so if you encounter connection issues, default to using this algorithm for most troubleshooting
scenarios.
Getting Started with SIP
To dial over SIP peer-to-peer without using a SIP server follow the instructions in this document. To
dial over SIP using a SIP Server you will need to:
1. Register the codec to a SIP server using SIP account credentials.
2. Configure a SIP interface in the codec. Note: The SIP1 or SIP2 interface will include the
proxy and port settings, as well as the selected IP interface used to make the connection,
e.g. LAN1.
Important Notes:
·SIP dialing is only supported over point-to-point connections, not multi-unicast
connections.
·Tieline G5 codecs support a SIP call being placed on-hold. Note: there are several
different implementations of "on-hold" by various SIP providers. Some will stop
streaming when a call is placed on-hold and others will replace live streaming with on-
hold messages or music.

8 Tieline G5 Codec Compatibility over IP and ISDN v1.2
© Tieline Pty. Ltd. 2019
·Tieline supports RFC5109 and RFC2733 compliant FEC over SIP from firmware
v2.18.xx.
·Some ISPs and/or cellular networks may block SIP traffic over UDP port 5060.
1.1 Connecting to a Comrex Access Rack Codec
Important Note:
·Firmware installed; Comrex Access codec firmware version 4.0 / Flash 4.0; Tieline
codec firmware version 2.14.88.
·To connect to the Comrex web-GUI you need to determine the IP address of the codec
on startup. View the PPM LEDs on start up to discover the last 4 digits of the IP
address; L(Send) is the 4th last digit and R(Receive) is the last digit.
·The Comrex Access does support MPEG 1 Layer 2 encoding.
·The default credentials for the Comrex Access GUI is to use any username and enter
the default password comrex
Common SIP settings Tieline recommends when connecting to non-Tieline codecs like Comrex are
as follows:
·Profile: Mono
·Bit rate: 64kbps
·Algorithm: G.722
·Session port: UDP 5060
·Audio port: UDP 5004
Configure the Comrex Access Rack Codec for a Peer-to-Peer SIP Call
1. Attach a LAN cable and any audio inputs to the Comrex Access codec.
2. Apply power to the Comrex Access.
3. Determine the IP address of the codec (DHCP by default), by noting the number of "Receive"
LEDs illuminated on the front panel of the Access codec.
4. Enter the IP address into a browser and open the Comrex Access Graphic User Interface (GUI).
5. Click the System Settings tab in the Comrex Access GUI.
6. Click to select the Show advanced options check-box below the System Setting pane.
7. Click the arrow symbol to the left of the EBU 3326/SIP Settings folder to expand and view all
SIP settings.
8. Click Accept incoming connections and ensure the Enabled check-box is selected.
9. Click IP Port and ensure that port 5060 is entered in the port number text box; click Apply to
change this setting after making changes.
10.Click RTP IP Port and ensure that port 5004 is entered in the port number box; click Apply to
change this setting after making changes.
11.Click the arrow symbol for Standard RTP Settings to expand and view all settings.
12.Click IP Port and change the Incoming network port to 5004, then click Apply.
13.Click the Profiles tab and click the Add New Profile button.
14.In the Profile Setting section select General and type a name into the Profile Name text box,
then select EBU 3326/SIP in the Channel drop-down arrow selections.
15.In the Profile Setting section and select Local, then Encoder and click VoIP in the
adjustment pane to select either G.711 or G.722. Then select Remote > Encoder > VoIP >
G.722. Note: Alternatively, select AAC, HE-AAC, AAC-LD or Opus encoding options.
16.You are now ready to connect, so click on the Connections tab in the Comrex Access GUI.
17.Click the Store New Remote button to create a new connection.
18.Enter the Remote Name and the IP address of the Tieline codec you are dialing. Then select
the profile you want to use via the drop-down Profile list box, next click OK.
19.Click the large green Connect button.
20.Click the Disconnect button to hangup the connection.

9Connecting Tieline IP to other Codecs Using SIP
© Tieline Pty. Ltd. 2019
Configure a Tieline G5 Codec for a peer-to-peer SIP Call
1. Attach a LAN cable and any audio connections to the Tieline codec.
2. To dial peer-to-peer press the HOME button to return to the Home screen, select
Connect > IP > SIP.
3. Use the numeric KEYPAD to enter the IP address of the codec you want to dial, using the
or buttons to enter the periods in the IP address and use the RETURN button
to delete numbers already entered.
4. Then press the down navigation button to select Setup and press to adjust the
algorithm to match the Access rack codec. Also adjust the jitter buffer and encode/decode
direction if required. Note: By default the Tieline codec will attempt to connect using MP2
and then G.722.
5. Press the RETURN button to navigate backwards to the Connect SIP screen.
6. Press the CONNECT button to make a connection.
Important Note: The Tieline codec should automatically select G.711 µ -law for North
America and Japan, and G.711 a-law in most other regions of the world if this algorithm is
used.
1.2 Connecting to a Comrex Access Portable
Important Note: Firmware installed - Comrex Access Portable codec firmware version
2.8-pre; Tieline codec firmware version 2.14.100
Configure the Access Portable for a Peer-to-Peer SIP Call
1. Attach a LAN cable and any audio inputs to the Comrex Access codec.
2. Apply power to the Comrex Access Portable.
3. Tap the Configure drop-down menu in the top-left of the codec touch-screen and then tap
System Settings.
4. Tap the Advanced check box at the bottom of the screen to display the full menu.
5. Scroll down the screen to N/ACIP SIP Settings and tap to select Accept Incoming
Connections if it is not enabled, then tap Edit, next tap the Enabled check box and tap the
Save button.
6. Ensure that port 5060 is entered into the port number box.
7. Tap RTP IP Port and enter port 5004 into the port number box.
8. Tap the Network drop-down menu at the top of the screen, then tap Manage Networks to
determine the IP address (DHCP by default) of the codec as you will need this to dial into the
codec from a Tieline codec.
9. Tap the Configure drop-down menu in the top-left of the codec touch-screen and then tap
Manage Profiles.
10.Tap the Add New button to create a New Profile in the Available Profiles list.
11.Tap New Profile to highlight it and tap the Edit button.
12.Tap Profile Name to highlight it on the screen and tap the Edit button, then type a name for the
profile and tap the Save button.
13.Tap Channel to highlight it on the screen and tap the Edit button, then tap to select N/ACIP
SIP in the drop-down list box and tap the Save button.
14.Tap Encoder to highlight it on the screen and tap the Edit button, then tap X3: VoIP G.722 in
the drop-down list box and tap the Save button.
15.Tap the Done button in the Profile Settings screen.
16.To create a new "Remote" connection tap Remotes, then tap Add New Remote.
17.Enter the Name of the connection and the IP address, then tap to select the profile you have
just created in the Profile drop-down list box, next tap the OK button.
18.Tap on the Remote you have just programmed with the new profile and tap the Connect button
on the screen to connect to the Tieline codec.

10 Tieline G5 Codec Compatibility over IP and ISDN v1.2
© Tieline Pty. Ltd. 2019
19.Tap the Disconnect button to hangup the connection.
Configure a Tieline G5 Codec to Connect peer-to-peer using SIP
1. Attach a LAN cable and any audio connections to the Tieline codec.
2. To dial peer-to-peer press the HOME button to return to the Home screen, select
Connect > IP > SIP.
3. Use the numeric KEYPAD to enter the IP address of the codec you want to dial, using the
or buttons to enter the periods in the IP address and use the RETURN button
to delete numbers already entered.
4. Then press the down navigation button to select Setup and press to adjust the
algorithm to match the Access portable codec. Also adjust the jitter buffer and
encode/decode direction if required. Note: By default the Tieline codec will attempt to
connect using MP2 and then G.722.
5. Press the RETURN button to navigate backwards to the Connect SIP screen.
6. Press the CONNECT button to make a connection.
Important Note: The Tieline codec should automatically select G.711 µ -law for North
America and Japan, and G.711 a-law in most other regions of the world if this algorithm is
used.
1.3 Connecting to a Mayah Sporty
Important Note: Mayah Sporty codec firmware version 4.9.1.0; Tieline codec firmware
version 2.14.88.
Configure a Mayah Sporty for a Peer-to-Peer SIP Call
1. Attach a LAN cable and any audio inputs to the Sporty codec.
2. Apply power to the Mayah Sporty.
3. Find the IP address of the codec via F2 [Codec1] > F3 [Setup] > F1 [Interface] > Ethernet >
OK. Use the navigation buttons to find the IP address as you will need this to dial into the codec
from the Tieline codec.
4. Use the up/down navigation arrow buttons and select SIP.
5. Navigate down and ensure port 5060 is entered for the UDP port, then ensure port 5004 is
entered for the TCP port. Select UDP or TCP and then OK to edit these settings.
6. Press the F4/ON button several times to return to the Home screen.
7. Press F2 [Codec] > F3 [Setup] > F2 [Quality] to display the algorithm selection screen.
8. Use the navigation arrow buttons on the top of the codec to select an algorithm from G.711µ ,
G.711a, G.722, MPEG L2 or Linear 16 bit.
9. Press the F4/ON button several times to return to the Home screen.
10.Ensure the codec is configured to automatically accept incoming calls via F4/ON System > F2
(Misc.) > Connections > Accept Mode > Auto.
11.Press the F4/ON button several times to return to the Home screen.
12.Select F2 [Codec1] > (F1) Connect > (F3) Direct > Interface (Ethernet) > Protocol (SIP) >
SIP Address (enter IP address) > OK.
13.Press the right navigation button to select dial and then press OK to dial. The screen should
display Connecting and then connect to the destination codec.
14.To hang up from the Home screen select F2 [Codec1] > F1 (End Call).
Configuring a Tieline G5 Codec for a Peer-to-Peer SIP Call
1. Attach a LAN cable and any audio connections to the Tieline codec.
2. To dial peer-to-peer press the HOME button to return to the Home screen, select
Connect > IP > SIP.