Codec Tieline G6 Codec SIP User manual

Tieline G6 Codec SIP
Compatibility over IP
Manual Version: 1.0
October, 2021

Tieline G6 Codec SIP Compatibility v12
© Tieline Pty. Ltd. 2021
Table of Contents
Part I Connecting Tieline to other
Codecs Using SIP 3
................................................................................................................................... 81 Connecting to a Comrex AccessRack Codec
................................................................................................................................... 82 Connecting to a Comrex AccessPortable
................................................................................................................................... 93 Connecting to a Mayah Sporty
................................................................................................................................... 104 Connecting to a TelosZephyr IP
................................................................................................................................... 105 Connecting to an APT Worldcast Equinox
................................................................................................................................... 116 Connecting to an ProdysProntonet LC
Part II Configure Tieline SIP Interfaces
and Accounts 12
................................................................................................................................... 121 Configuring SIP Interfaces
................................................................................................................................... 132 Configuring SIP Accounts
................................................................................................................................... 163 Configure SIP Allow and Block Lists
Part III Configure Peer-to-Peer SIP
Programs 18

3Connecting Tieline to other Codecs Using SIP
© Tieline Pty. Ltd. 2021
1 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 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 G6 Codec SIP Compatibility v1
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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. 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.
3. SIP connections are more complex to configure.

5Connecting Tieline to other Codecs Using SIP
© Tieline Pty. Ltd. 2021
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 by external
parties.
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.
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 Allow and Block lists" in the product
user manual 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" in the product user
manual for more information.
6. Firewall settings facilitate enabling or disabling a range of firewall-related network services, or
limit ping to only work in a local subnet. Tieline also recommends SNMP is disabled if a
codec is connected to a public network like the internet. Adjust settings using the Toolbox
HTML5 Web-GUI Options panel in the Firewall tab, or see "Firewall Configuration" in the
product user manual.
7. Implementation of CSRF protection (Cross-Site Request Forgery). Enable and disable this
setting using the Options panel in the Toolbox HTML5 Web-GUI, or see "Enabling CSRF
Security" in the product user manual 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 G6 Codec SIP Compatibility v1
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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 allow or block
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.
How do I find myCodec's User Agent
It may be necessary to identify a codec user agent as some codec manufacturers allow calls based
on 'User Agent' identification. 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 the user agent in the codec is configured as "Tieline <Product Name>
<Firmware Version>". E.g. Tieline Bridge-IT v2.18.68
·User agent for Report-IT SIP connections: Tieline Report-IT EE (3.5.6_2894). Note: In this
example "3.5.6" is the Report-IT version number and "2894" is the build number.
·In Tieline G3 codecs the user agent is configured as "Tieline <Product Name> <Serial
Number>". E.g. Tieline TLR350 8972. The model numbers for Tieline G3 codecs are as
follows:
oCommander G3 Rack TLR300 = Model Number TLR300
oCommander G3 Rack TLR300B = Model Number TLR350
oCommander G3 Field TLF300 = Model Number TLF300
oi-Mix G3 TLM600 = Model Number TLM600

7Connecting Tieline to other Codecs Using SIP
© Tieline Pty. Ltd. 2021
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.
Using this Document
To dial over SIP peer-to-peer without using a SIP server follow the instructions in this document. The
following sections explain:
1. How to configure a range of codecs from different vendors to connect with Tieline G6 codecs.
2. How to configuring Tieline G6 codecs for SIP using the Toolbox Web-GUI.
If you require more detailed information about configuring a Tieline codec for SIP, visit
www.tieline.com/support and download the user manual for your product.
Notes on Using Comrex Codecs
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
Please note the following when connecting to Comrex codecs using SIP:
1. Some Comrex codecs do not support MPEG Layer 2, so G.722 is the preferred option for
testing.
2. Opus encoding may be the default option in some Comrex codecs. Current Tieline firmware
supports both CBR and VBR Opus encoding. In BRIC-Link II, and possibly other codecs,
special CBR modes are offered for compatibility with Tieline products if firmware in a Tieline
codec is old, i.e. prior to version 2.18.32.
3. Comrex codecs using SIP may expect a User Agent ID. Please see "How do I find my
Codec's User Agent" in this document for more details on Tieline codec User Agent IDs.

8 Tieline G6 Codec SIP Compatibility v1
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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 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.
·The default credentials for the Comrex Access GUI is to use any username and enter the
default password comrex
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.
1.2 Connecting to a Comrex Access Portable
Important Note:
·Firmware installed - Comrex Access Portable codec firmware version 2.8-pre.
·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.
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.

9Connecting Tieline to other Codecs Using SIP
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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.
19.Tap the Disconnect button to hangup the connection.
1.3 Connecting to a Mayah Sporty
Important Notes:
·Mayah Sporty codec firmware version 4.9.1.0
·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.
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.

10 Tieline G6 Codec SIP Compatibility v1
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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).
1.4 Connecting to a Telos Zephyr IP
Important Notes:
·Zephyr IP codec firmware version 3.1;
·The default user name "user" and password "Telos" can be used to open the Telos web-
GUI interface.
·In testing Tieline was successful in connecting using MPEG Layer 2 and G.722.
Configure the Zephyr IP for a Peer-to-Peer SIP Call
1. Apply power to the codec and when the menu appears use the arrow buttons to navigate to
Network on the LCD screen and press OK.
2. Verify and note the IP address displayed via Ethernet Config.
3. Press the ESC button and return to the Main Menu Screen.
4. Select Codec > Advanced Setup > Encoding Mode [Layer 2] / Minimum Bitrate
[128kbps] / Maximum Bitrate [128kbps]. Note: G.722 has also been successfully tested.
5. To add a new sip address navigate to Auto > Add > Device Name [sip1@<enter IP address
here>] >Device Type [SIP] > Save. Note: the @ symbol is accessed via the "1" button.
6. Navigate back to the Contacts screen, select the contact and then select Call and press OK to
dial.
Important Notes: The address used to dial the Zephyr from the Tieline codec over SIP was
ZEPHYR@<insert IP address here>
1.5 Connecting to an APT Worldcast Equinox
Important Notes:
·Equinox codec firmware version 3.1;
·In testing Tieline was successful in connecting using MPEG 1 Layer 2 encoding only.
Configure the Equinox for a Peer-to-Peer SIP Call
1. Plug your Ethernet LAN cable into the back of the codec and attach power.
2. Ensure the correct IP address is configured in the Equinox via Main Menu > IP > Stream
Port Settings.
3. Return to the Main Menu and select Audio.
4. Next select Audio Profile (No) and then MPEG - L2 as the algorithm. Select the
appropriate bit rate and whether you want to dial in mono, stereo or Joint Stereo, and then
the sample rate. For the profile we selected CCS IMUX.
5. Return to the Main Menu and select User and navigate to Primary Conn., then press the
Ent Dial button.
6. Navigate to Codec - SIP > Master and press Ent Dial.
7. Return to the Main Menu and select the SIP menu and press the Ent Dial button.
8. Select Setup Address Book and press the Ent Dial button.
9. Select entry entry 0or 1in the address book and press the Ent Dial button to enter the
address. Note: if you use entry 0or 1you can use the FD0 and FD1 buttons on the front of
the codec as speed dial buttons for these two entries.
10.Configure the address as SIP:TIELINE@<insert codec IP address>.
11.The codec should now be ready to dial or answer.
12.Press either the FD0 or FD1 button to dial one of these entries.

11Connecting Tieline to other Codecs Using SIP
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Important Notes:
·The default Local SIP address in the Equinox you need to dial is
sip:[e-mail] . As an example:
default_local_sip_user@203.36.205.174
·You can adjust the local SIP address setting in the Equinox via Main Menu Screen >
SIP > Local User Address
1.6 Connecting to an Prodys Prontonet LC
Important Notes:
·Prodys Prontonet codec firmware version 06.0.1
·In Tieline testing the codecs were successful in connecting over MPEG Layer 2 in both
directions.
·Please configure the Prontonet LC with a 20ms frame rate when attempting to connect
over G.711 or G.722 with Tieline codecs. This setting can be configured using Prodys
web-browser configuration software.
·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.
Configure a Prodys Prontonet LC for a Peer-to-Peer SIP Call
1. Attach a LAN cable and any audio inputs to the Prontonet codec.
2. Apply power to the codec.
3. To find the IP address in the codec press the INF button on the codec until you can see the IP
address listed adjacent to ADDR:
4. Press the OK button to exit the INF menu screen.
5. Press OK again to view the MAIN MENU.
6. Use the navigation buttons to select NET and press OK.
7. In the NET SELECTION screen select IP and press OK.
8. In the SET CODEC screen select SIMPLE for a single connection, then press OK.
9. In the SET IP NET screen select OK when Unicast/Multicast is displayed.
10. In the STREAMING PROTOCOL screen navigate to SIP and press OK.
11.Press arrow button above the OK button to return to the MAIN MENU.
12.Navigate to ENC and press OK.
13.Press OK in the SET ENCODING MODE (ENCODER1) screen.
14.In the SET ENCODER 1 screen navigate to your preferred encoding algorithm and press OK.
15.Configure your preferred bit rate in the MPEG SET BITRATE screen, then press OK.
16.Select Mono, J-Stereo or Stereo in the MPEG SET MODE screen, then press OK.
17.Select your preferred sample rate in the MPEG SET Fs screen, then press OK.
18.In the MPEG SET CRC menu select OFF and press OK.
19.In the MPEG SET AUX DATA menu select OFF and press OK.
20.Press the CALL 1 button and select Unicast, then press OK.
21.Select BIDIR for a bidirectional audio path over the connection, then press OK.
22. In the LAN L1 DIAL screen use the keypad buttons to enter the IP address of the destination
codec in the LAN L1 DIAL screen, then press OK to dial.
23.On the LCD screen of the Prontonet it should display L1->CONNECTED and FRAMED is
displayed in the bottom left corner of the screen. The GREEN LED adjacent to the CALL1
button is also illuminated when connected.
24.Press the CALL 1 button for approximately 2-3 seconds to hangup the call.

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2 Configure Tieline SIP Interfaces and Accounts
Tieline G6 codecs support dialing over SIP using a registered SIP server account, or peer-to-peer
using an available SIP interface, e.g. SIP1 or SIP 2. To configure a peer-to-peer SIP connection see
Configure Peer-to-Peer SIP Programs.
To dial over SIP using a SIP Server you will first 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.
3. Create a SIP program using the HTML5 Toolbox Web-GUI.
Important Notes:
·SIP dialing is only supported over point-to-point connections, not multi-unicast
connections.
·Failover and SmartStream PLUS redundant streaming is not available with SIP.
·Tieline supports RFC5109 and RFC2733 compliant FEC over SIP from firmware v2.18.xx.
·Tieline G5 and G6 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.
·Some ISPs and/or cellular networks may block SIP traffic over UDP port 5060.
·By default the Tieline codec will attempt to connect using MP2 and then G.722
2.1 Configuring SIP Interfaces
Important Notes:
1. SIP interfaces are disabled by default.
2. SIP1 is configured to use LAN1 by default, which is mapped to the Primary Via
interface by default.
3. SIP2 is configured to use LAN2 by default, which is mapped to the Secondary Via
interface by default.
4. SIP1 and SIP2 each need to use a separate IP interface when connecting, e.g. LAN1 or
LAN2.
5. SIP1 and SIP2 can each make multiple SIP calls, e.g. two calls can be made over
SIP1, or two calls can be made over SIP2.
6. The settings for SIP1 and SIP2 cannot be edited if the interface has been enabled.
7. Enter a public IP address in the Public IP text box if you want to dial over SIP from
behind a firewall. Then configure port forwarding to route traffic to the codec's local IP
address behind your firewall. Note: Do not enter a Public IP address if STUN is
configured. They cannot be used together because both will attempt to use a public IP
address over SIP. STUN settings are prioritized and used if both are configured.
To configure SIP1 or SIP2:
1. Open the HTML5 Toolbox Web-GUI and click Transport and then click SIP Interfaces to view
and configure SIP interface settings.
2. Default SIP settings are configured and select Interface SIP1 or Interface SIP2 to adjust each
interface. Note: Ensure each interface uses a unique "Via" IP interface because they can't share
one, e.g. LAN1.

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3. Select the Enable check-box and then click the Save button to confirm settings.
4. The SIP interface indicator is green when an interface is enabled and red when it is disabled.
2.2 Configuring SIP Accounts
Up to 16 SIP accounts can be configured in the codec and registering codecs for SIP connectivity
is simple. First, select the SIP server to which you will register your codec. On a LAN this may be
your own server, or it could be one of the many internet servers available. When you register an
account with a SIP server you will be provided with:
·Username
·Authorized User
·SIP address
·Domain
·Realm
·Registrar
·Registar port
·Outbound Proxy
·Proxy port
·Password
·Registration Timeout (this shouldn't need to be adjusted from the default setting).

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Important Notes:
·In most situations it is best to configure a SIP account when the codec is configured with
a public IP address.
·Each SIP account can only be mapped to a single SIP interface, i.e. SIP1 or SIP 2.
·Up to 16 SIP accounts can be added to the codec.
Adding a SIP Account
Enter SIP account details and register the account in your codec. Once configured, the codec will
contact the SIP server automatically to acknowledge its presence over a wide area network when
connected to a public IP address.
1. Open the HTML5 Toolbox Web-GUI and click Transport and then click SIP Accounts to
view and configure SIP account settings.
2. Click to select one of the unused Accounts at the top of the SIP Accounts panel.
3. Enter the SIP account details into the relevant text boxes, including the registration
Timeout (which shouldn't need to be adjusted from the default setting). Also ensure a SIP
Interface is selected (e.g. SIP1 or SIP2.) The SIP interface contains settings related to
ports and the selected Via interface, e.g. LAN1 or LAN2.
4. Click the Enable check-box at the top of the panel and then click the Save button to
register the codec to the server.
5. If an account is registered successfully, the account registration indicator changes from red
to green, and Not Registered (above the Enable check-box) becomes Registered.

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6. In the Toolbox Web-GUI the red SIP indicator adjacent to the codec Online indicator also
changes to green when an account is currently registered in the codec and ready to be used
when dialing over SIP.
7. Once enabled, the SIP account can be selected when creating a new SIP connection.
Important Notes:
·Some ISPs may block SIP traffic over UDP port 5060.
·By default, the session port used for each SIP account is the associated SIP interface
session port. The default session port is the registered UDP port number 5060. It is also
possible to configure a custom local session port for each SIP account for compatibility
with Cisco Unified Communications Manager (CUCM). Ensure your firewall has the
required TCP and UDP ports open when receiving multiple SIP calls.
Troubleshooting SIP Registration
If a SIP account is not registering successfully please check the following:
1. Confirm all account registration information has been entered correctly.
2. Confirm the SIP interface (SIP1 or SIP2) configured as the Via in the account is enabled.
3. Verify that the Via selection in the SIP1 or SIP2 interface settings corresponds with the
network interface being used by the codec to register the account. E.g. LAN1, LAN2.

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2.3 Configure SIP Allow and Block Lists
The SIP Filter Lists panel allows filtering of SIP URIs and User Agents to provide greater security
for connections. For example, add trusted network codecs to the URI Allow List in this panel and
only codecs using these SIP URIs will be able to connect. This is like saying, "if you have the key
you can open the door" and is perhaps the easiest way to filter outside access to your codec's "front
door".
It is also possible to add SIP URIs to the URI Block List and add user agents to the User Agent
Block List to deny them access to the codec. These block lists also filter unwanted traffic and
increase the likelihood of rejecting unwanted traffic. Note: If an incoming SIP caller is not on the URI
Allow List it will be scanned using the URI Block List. If there is no match it will be scanned using
the User Agent Block List. A connection will be established if there is no match on either Block
List.
Important Note: To only allow a predefined list of codecs to connect, add them to the URI
Allow List and add a wildcard (asterisk) * to the URI Block List: all incoming calls will be
blocked except for codecs in the Allow List.
Filter URIs and User Agents
1. Open the HTML5 Toolbox Web-GUI and click Transport in the Menu Bar, then click SIP Filter
Lists to launch the SIP Filter Lists panel.
2. Click the Plus symbol for URI Allow List, URI Block List or User Agent Block List to add
a new item to the list.
3. Enter the new item in the text box, click to select the check-box and then click Save to store
the new setting.
4. Click the Undo symbol to undo editing and click and drag the List symbol to shift the
position of allow list and block list items.
Important Note: Some codec manufacturers allow calls based on 'User Agent'
identification. It may be necessary to enter a Tieline codec user agent into a non-Tieline
codec to connect to a Tieline SIP-enabled codec.
·From firmware v2.16.xx the user agent in the codec is configured as "Tieline <Product
Name> <Firmware Version>". E.g. Tieline Bridge-IT v2.18.68
·User agent for Report-IT SIP connections: Tieline Report-IT EE (3.5.6_2894). Note: In this
example "3.5.6" is the Report-IT version number and "2894" is the build number.
·In Tieline G3 codecs the user agent is configured as "Tieline <Product Name> <Serial
Number>". E.g. Tieline TLR350 8972. The model numbers for Tieline G3 codecs are as
follows:

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oCommander G3 Rack TLR300 = Model Number TLR300
oCommander G3 Rack TLR300B = Model Number TLR350
oCommander G3 Field TLF300 = Model Number TLF300
oi-Mix G3 TLM600 = Model Number TLM600
Using Regular Expressions
To filter using regular expressions in the SIP Filter Lists panel, click the Options symbol in the
top right-hand corner of the panel and then click to select the Use Regular Expressions check-
box.
Important Note: Regular expressions should not use ^ and $ anchors because searches
implicitly try to match anywhere in the line.

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3 Configure Peer-to-Peer SIP Programs
The codec supports dialing over SIP using a registered SIP server account, or peer-to-peer using
one of the two SIP interfaces SIP1 and SIP 2. It is also necessary to select SIP as the Session
Protocol.
To configure a SIP multiple stream program simply create a new program and configure each SIP
audio stream like a single SIP Peer-to-Peer program. The codec is capable of registering up to 16
SIP accounts, each of which has an associated Answer Route field, which can be matched to a
loaded answering program's audio stream Answer Route. Without using SIP accounts, each SIP
interface also has an Answer Route field. However, only 2 SIP interfaces are supported, limiting
this method of routing configuration to a maximum of 2 audio streams. Note: An account's Answer
Route setting is applied first.
Important Notes: Before commencing program configuration please note:
·You cannot edit a program when it is currently loaded in the codec.
·Some drop-down menus and settings may be greyed out intentionally depending on
features available.
·Failover and SmartStream PLUS redundant streaming is not available when
connecting using SIP.
·Lock a loaded custom program or multistream program in a codec to ensure it cannot
be unloaded by a codec dialing in with a different type of program. For example, if a
multistream program is not locked it will be unloaded by a mono or stereo call.
·Ensure the appropriate TCP and UDP audio ports are open in your firewall to allow SIP
audio streams to connect.
1. Open the HTML5 Toolbox Web-GUI and click Connect in the Menu Bar, then select Program
Manager to launch the Program Manager panel.
2. Click the New Program button to open the wizard and:
·Click in the text box to name the new program.
·Click the Mix drop-down arrow to associate a custom matrix mix with the program if required.
·Select Mono/Stereo Peer-to-Peer, or if you want to use an existing program as a template,
select this option. Then click Next.
Important Notes: When you use an existing program as a template, the new program
inherits all the settings of the template program and you can adjust these settings as
required by continuing through the program wizard.
3. To configure new program level rules click the drop-down arrow and select the preferred
option from those available. Click the blue Plus symbol to add a new rule and click the
Minus symbol to remove a rule.

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Important Notes for Rules:
·The Gateway 4 codec has 8 hardware GPIOs and 56 logical outputs, and the Gateway
8/16 has 16 hardware GPIOs and 48 logical outputs; both codecs also have 3 virtual
inputs, and 64 WheatNet Logic Inputs/Outputs. (WheatNet logic I/Os allow Tieline
WheatNet-IP enabled codecs to activate functions across a WheatNet-IP network.
WheatNet logical inputs are only available if a codec has a WheatNet-IP card installed).
·A non-WheatNet-IP Tieline codec can be configured to trigger a WheatNet LIO in a Tieline
WheatNet-IP codec.
·Tieline WheatNet-IP codecs require Wheatstone Razor firmware version 1.4.22 or later to
support WheatNet LIOs. In addition, the WheatNet-IP codec must have the WNet Enable
LIO checkbox selected in the Options panel of the HTML5 Toolbox Web-GUI.
·Relay reflection is not available for SIP and Multicast Client programs.
·For more details about rules see download the product user manual at
www.tieline.com/support.
4. Enter the Stream Name and configure the codec to dial, answer or dial and answer. Then
click Next.
Note: The following example will display how to configure a dial and answer program. If you want the
codec to either dial or answer only, select the option and the wizard will automatically display
screens to allow you to configure the codec correctly. Please note that caller ID, dial routes,
TieLink, and G3 profile or G3 channel information can not be used for SIP connections because
Tieline session data is replaced by SDP for SIP connections.
5. This audio stream connection in the wizard will allow the codec to dial. Enter the name of
the connection in the text box, then click Next.
6. Configure the transport settings for the connection. Ensure that you select:
·IP as the Transport.
·SIP from the Session Protocol menu option.

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Then click Next.
7. Configure the destination codec Address if you are dialing peer-to-peer, then specify the
network interface used to dial the connection, e.g. Primary (Ethernet port 1). Enter the
name of a registered SIP account if you are using a SIP server to establish a connection. If
you wish to dial from one of the codec's registered accounts, then enter the account name in
the Account field using the format accountname@sipserverdomain, e.g.
[e-mail]. In this configuration the account interface will be used rather
than the specified Via, e.g. if the account is using SIP2 and this is configured to use LAN2,
then the call will proceed using LAN2. If you do not wish to use an account for the dial then
leave the Account field blank and select the required interface. Note: the interface must be
associated with either SIP1 or SIP2 for the call to be able to proceed.
At this point you can click Save Program and save the program with default algorithm and jitter
settings. Alternatively, click Next to confirm and specify algorithm and jitter settings for this
connection and configure backup audio settings (recommended).
Important Notes:
·The default UDP audio port when using SIP for a peer-to-peer connection is 5004 in
Tieline codecs. To contact a codec that is behind a firewall or NAT-enabled router, it is
essential that this and all other relevant ports are open and forwarded to the other device.
·Tieline codecs automatically add "sip:" to the address you enter in the Address field
when dialing, so it's not necessary to add this.
·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 5060.
8. Click the drop-down arrows on the right-hand side of each active drop-down menu to adjust
the Encoding, Sample rate or Bit rate parameters. Click Next to continue.