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21

Configure Peer-to-Peer SIP Programs

© Tieline Pty. Ltd. 2021

9.

 

Click to configure:

·

Auto  Jitter  Adapt

  and  the  preferred  auto  jitter  setting  using  the  drop-down  arrow  for 

Buffer

priority

.  It  is  also  possible  to  configure  the 

Minimum  depth

  and 

Maximum  depth

  of  jitter

over the connection.

·

Alternatively,  select  a 

Fixed  Buffer  Level

  and  enter  the 

Jitter  Depth

,  which  must  be

between 12ms and 5000ms depending on the algorithm you select.

·

RFC-compliant FEC can also be configured if required 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  no  guarantee  that  the
remote  codec  will  accept  or  support  these  jitter  configuration  settings.  Verify  configuration
settings  on  the  remote  codec  to  ensure  settings  are  correct.  Recommended  jitter  buffer
limits are as follows:

·

1,000ms for PCM and G.711, G.722 and aptX Enhanced encoding.

·

2,500ms for AAC ELD, AAC LD.

·

5,000 for  all  other  algorithms  including  Opus,  MP2,  AAC,  AAC-HE,  Tieline  Music  and  Music
PLUS.

11. Click 

Next

 to select the check-box if you want to 

Enable Auto Reconnect

.

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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