Codec Tieline G6 Codec SIP Скачать руководство пользователя страница 11

11

Connecting Tieline to other Codecs Using SIP

© Tieline Pty. Ltd. 2021

Important Notes:

 

·

The default Local SIP address in the Equinox you need to dial is

sip:[email protected] . 

As an example:

[email protected]

·

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.

Содержание Tieline G6 Codec SIP

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

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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...

Страница 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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