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10

Tieline G6 Codec SIP Compatibility v1

© Tieline Pty. Ltd. 2021

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 

0

 or 

1

 in the address book and press the 

Ent Dial

 button to enter the

address. Note: if you use entry 

0

 or 

1

 you 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.

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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