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Tieline G6 Codec SIP Compatibility v1

© Tieline Pty. Ltd. 2021

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

Summary of Contents for Tieline G6 Codec SIP

Page 1: ...Tieline G6 Codec SIP Compatibility over IP Manual Version 1 0 October 2021...

Page 2: ...to a Comrex Access Portable 9 3 Connecting to a Mayah Sporty 10 4 Connecting to a Telos Zephyr IP 10 5 Connecting to an APT Worldcast Equinox 11 6 Connecting to an Prodys Prontonet LC Part II Configu...

Page 3: ...s 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 tw...

Page 4: ...ss 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...

Page 5: ...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...

Page 6: ...y 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 i...

Page 7: ...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 S...

Page 8: ...at 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...

Page 9: ...p 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 Remo...

Page 10: ...lect 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 Importan...

Page 11: ...se 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...

Page 12: ...rks 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...

Page 13: ...e 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 on...

Page 14: ...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 ac...

Page 15: ...ssion 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 Ens...

Page 16: ...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 M...

Page 17: ...Number TLF300 o i 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 o...

Page 18: ...e 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...

Page 19: ...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 Ente...

Page 20: ...e 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 Alte...

Page 21: ...quired and the percentage is configurable 10 Click Add a remote jitter preference to send preferred jitter settings to a remote codec Note this is just a preference as per EBU Tech 3368 and there is n...

Page 22: ...possible to configure remote jitter preferences if the remote codec supports RFC5109 15 Click Next to configure Failure Parameters for the answering connection if required Please note In most situatio...

Page 23: ...m a remote codec Note this must be selected as one of the configured sources Input Input audio looped to the physical codec outputs HTTP Icecast client mode to allow media server streaming from a spec...

Page 24: ...the blue Plus symbol to add a new rule and click the Minus symbol to remove a rule Important Note Program level rules intended to activate dialing are not valid in Answer only programs or audio strea...

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