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Difference between revisions of "Multicast Paging (SIP)"

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'''Multicast Paging''' enables IP intercom stations in SIP mode to receive VoIP audio as multicast paging from 3rd party iPBX (e.g. Asterisk). The feature is supported by both [[:Category:Stations#INCA_stations|INCA stations]] and [[:Category:Stations#Turbine_stations|Turbine stations]]. Up to 10 different Multicast paging groups can be defined.
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'''Multicast Paging''' enables IP intercom stations in SIP mode to receive VoIP audio as multicast paging from 3rd party iPBX (e.g. Asterisk). The feature is supported by both [[:category:INCA Stations|INCA stations]] and [[:category:Turbine|Turbine stations]]. Up to 10 different Multicast paging groups can be defined.
 
 
{{note|
 
{{note|
 
* ''Multicast Paging is supported in '''SIP mode''' only''
 
* ''Multicast Paging is supported in '''SIP mode''' only''
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* ''Required software: Turbine 4.9 or newer. INCA 2.9 or newer'' }}
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* ''Required software: [[:Category:Stations#Turbine_stations|Turbine]] 4.9 or newer. [[:Category:Stations#INCA_stations|INCA]] 2.8 or newer'' }}
  
 
== Configuration ==
 
== Configuration ==
 
To configure a station to receive Multicast Paging, [[Logging into an IP Station|log into the IP Station]] and select '''SIP Configuration''' > '''Multicast Paging''':
 
To configure a station to receive Multicast Paging, [[Logging into an IP Station|log into the IP Station]] and select '''SIP Configuration''' > '''Multicast Paging''':
  
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[[File:SIP Multicast.PNG|thumb|left|500px|Configuration page for Multicast Paging]]
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<table style="max-width:80%; border-style: double; border-color: #c7c7c7;">
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<br style="clear:both;" />
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<tr style="rowspan:2;">
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<td style="display: block; vertical-align: bottom;">[[Image:SIP Multicast.PNG|500px]]</td>
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</tr>
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<tr>
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<td style='background-color:#efefef;'>Configuration page for Multicast Paging</td>
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</tr>
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</table>
  
 
* '''Enable Multicast paging''': When enabled, the station is capable of playing audio received as Multicast
 
* '''Enable Multicast paging''': When enabled, the station is capable of playing audio received as Multicast
 
* '''Enable Order Priority''':  
 
* '''Enable Order Priority''':  
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** '''Enabled''': If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same priority, the VoIP stream with the highest "Order" number will be played.  
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** '''Enabled''': If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same "''Priority''", the VoIP stream with the lowest "''Order''" priority number will be played.  
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** '''Disabled''': The "Order" number will be ignored. If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same priority, the first received VoIP stream will be played. Subsequent VoIP streams will not be played.
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** '''Disabled''': The "''Order''" will be ignored. If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same priority, the first received VoIP stream will be played. Subsequent VoIP streams will not be played.
  
 
* '''Label''': Any descriptive text
 
* '''Label''': Any descriptive text
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* '''Listening Address''': The multicast IP address that the station shall liten to, e.g. 239.195.124.240. Multicast IP addresses are defined to be in the range from 224.0.0.0 to 239.255.255.255.
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* '''Listening Address''': The multicast IP address that the station shall listen to, e.g. 239.195.124.240. Multicast IP addresses are in the range from 224.0.0.0 to 239.255.255.255.
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* '''Port''': The port number must be within the port range 61020-61250, recommended range is 61080-61250. Using odd number for port is not recommended.  
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* '''Port''': The ports must be within the VoIP port firewall range, recommended range is 61080-61250 to avoid conflict with normal audio operation. Using odd numbers is not recommended for ports as that is normally used for RTCP. Listening address and port must be defined, and the port must be unique in the list.
 
* '''Codec''': Select between codecs G.711a, G.711u, G.722 and G.729
 
* '''Codec''': Select between codecs G.711a, G.711u, G.722 and G.729
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* '''Priority''': Select which priority mechanisms should apply to this multicast VoIP stream. Options are LOW, NORMAL, HIGH (default), and EMERGENCY. See more details below.
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* '''Priority''': Select which priority mechanism should apply to this multicast VoIP stream. Options are LOW, NORMAL, HIGH (default), and EMERGENCY. See more details below.
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* '''Order''': If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same priority, the VoIP stream with the highest "Order" number will be played. Requires that the "''Enable Order Priority''" option is enabled.
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* '''Order''': If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same "''Priority''", the VoIP stream with the lowest "''Order''" priority number will be played. Requires that the "''Enable Order Priority''" option is enabled.
  
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{{note|'''Order''' number 1 has the highest priority, 10 has the lowest priority.}}  
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{{note|'''Order''' number 1 has the highest priority, 10 has the lowest priority.}}
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== Paging priority ==
 
== Paging priority ==
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The priority mechanism has the following properties:
 
The priority mechanism has the following properties:
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* A multicast page will terminate any ongoing page with lower priority. If two paging calls have same priority, then the one with highest order number gets priority.
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* A multicast page will terminate any ongoing page with lower priority. If two paging calls have same priority, then the one with lowest order number gets priority.
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* '''EMERGENCY''' priority includes a [[Volume and handset override (Pulse/SIP)|Volume and handset override]] feature:
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* '''EMERGENCY''' priority includes a [[Volume and handset override (SIP)|Volume and handset override]] feature:
 
** The volume is turned to maximum level on all receiving stations, and the audio is played on the loudspeaker, even if the handset is off-hook.
 
** The volume is turned to maximum level on all receiving stations, and the audio is played on the loudspeaker, even if the handset is off-hook.
 
** During emergency page the station will queue all incoming calls. When emergency page ends, the station will accept calls again.  
 
** During emergency page the station will queue all incoming calls. When emergency page ends, the station will accept calls again.  
 
** If a station sends a busy override to a station listening on emergency page, then emergency page will stop
 
** If a station sends a busy override to a station listening on emergency page, then emergency page will stop
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* If a station is in a regular point-to-point call when receiving a page of defined priority:
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* If a station is in a regular point-to-point call when receiving a multicast paging of defined priority:
 
** '''LOW / NORMAL''' priority: The paging audio is not played
 
** '''LOW / NORMAL''' priority: The paging audio is not played
 
** '''HIGH''' priority: The paging audio is mixed with the normal call
 
** '''HIGH''' priority: The paging audio is mixed with the normal call
 
** '''EMERGENCY''' priority: The regular call is muted
 
** '''EMERGENCY''' priority: The regular call is muted
  
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== Configuration of Multicast streaming sources ==
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=== Stream from FFMPEG ===
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Stream radio source in g722:
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{{Code|<nowiki> ffmpeg -re -i "http://lyd.nrk.no/nrk_radio_p13_mp3_h" -filter_complex aresample=16000,asetnsamples=n=160 -ac 1 -acodec adpcm_g722 -f rtp rtp://239.1.20.5:61111</nowiki>}}
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 +
 
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Stream radio source in g711u/pcmu:
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{{Code|<nowiki>ffmpeg -re -i "http://lyd.nrk.no/nrk_radio_p13_mp3_h" -filter_complex aresample=8000,asetnsamples=n=160 -ac 1 -acodec pcm_mulaw -f rtp rtp://239.1.20.4:61110</nowiki>}}
  
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== so so ==
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=== Stream from VLC===
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=== Stream from FFMPEG ===  
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To stream from VLC does not work due to variable payload size, same problem as on SNOM (Not possible to fix this). http://wiki.snom.com/Category:HowTo:Multicast_Audio#Multicast_via_VLC
  
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=== Stream from VLC===
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<br>
  
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=== Listen on Pulse Group Call===  
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=== Listen to Edge Group Call===
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Go to [[Group_Call_(Edge)#Configuration_of_group_calls|Edge Controller -> Group Calls]] and use the multicast address/port written in each group call. Note that the clients making the group call must use same codec as chosen for multicast paging.
  
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=== Use Multicast Paging with Asterisk===  
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<br>
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=== Listen to Pulse Group Call===
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Go to [[Group_Call_(Pulse)#Configuration_of_group_calls|Pulse Server -> Group Call]] and use the multicast address/port written in each group call. Note that the clients making the group call must use same codec as chosen for multicast paging.
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<br>
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=== Use Multicast Paging with Asterisk===
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Call 8998 to speak on multicast address:
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{{Code|<nowiki>exten => 8998,1,Dial(MulticastRTP/basic/239.1.2.4:61101//c/g722))</nowiki>}}
  
 
=== Use Multicast Paging with Freeswitch iPBX===  
 
=== Use Multicast Paging with Freeswitch iPBX===  
  
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<extension name="rtp_multicast_page">
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{{Code|<nowiki><extension name="rtp_multicast_page">
 
   <condition field="destination_number" expression="^pagegroup$|^7243$">
 
   <condition field="destination_number" expression="^pagegroup$|^7243$">
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     <action application="answer"/>
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     &#60;action application="answer"/&#62;
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     <action application="esf_page_group" data="239.1.20.5 61082"/>
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     &#60;action application="esf_page_group" data="239.1.20.5 61082"/&#62;
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     <action application="esf_multicast_write_codec" data="G722"/>
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     &#60;action application="esf_multicast_write_codec" data="G722"/&#62;
 
   </condition>
 
   </condition>
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  </extension>
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  </extension></nowiki>}}
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[[Category: SIP intercom - Configuration]]
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[[Category: Audio Settings and Audio Features]]

Latest revision as of 15:49, 25 October 2023

SIP Icon 300px.png

Multicast Paging enables IP intercom stations in SIP mode to receive VoIP audio as multicast paging from 3rd party iPBX (e.g. Asterisk). The feature is supported by both INCA stations and Turbine stations. Up to 10 different Multicast paging groups can be defined.

Note icon
  • Multicast Paging is supported in SIP mode only
  • Required software: Turbine 4.9 or newer. INCA 2.8 or newer


Configuration

To configure a station to receive Multicast Paging, log into the IP Station and select SIP Configuration > Multicast Paging:

SIP Multicast.PNG
Configuration page for Multicast Paging
  • Enable Multicast paging: When enabled, the station is capable of playing audio received as Multicast
  • Enable Order Priority:
    • Enabled: If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same "Priority", the VoIP stream with the lowest "Order" priority number will be played.
    • Disabled: The "Order" will be ignored. If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same priority, the first received VoIP stream will be played. Subsequent VoIP streams will not be played.
  • Label: Any descriptive text
  • Listening Address: The multicast IP address that the station shall listen to, e.g. 239.195.124.240. Multicast IP addresses are in the range from 224.0.0.0 to 239.255.255.255.
  • Port: The ports must be within the VoIP port firewall range, recommended range is 61080-61250 to avoid conflict with normal audio operation. Using odd numbers is not recommended for ports as that is normally used for RTCP. Listening address and port must be defined, and the port must be unique in the list.
  • Codec: Select between codecs G.711a, G.711u, G.722 and G.729
  • Priority: Select which priority mechanism should apply to this multicast VoIP stream. Options are LOW, NORMAL, HIGH (default), and EMERGENCY. See more details below.
  • Order: If the station is exposed to two (or more) multicast VoIP streams at the same time, and the VoIP streams have the same "Priority", the VoIP stream with the lowest "Order" priority number will be played. Requires that the "Enable Order Priority" option is enabled.
Note icon Order number 1 has the highest priority, 10 has the lowest priority.


Paging priority

Multicast paging uses 4 different priorities: LOW, NORMAL, HIGH, and EMERGENCY.

The priority mechanism has the following properties:

  • A multicast page will terminate any ongoing page with lower priority. If two paging calls have same priority, then the one with lowest order number gets priority.
  • EMERGENCY priority includes a Volume and handset override feature:
    • The volume is turned to maximum level on all receiving stations, and the audio is played on the loudspeaker, even if the handset is off-hook.
    • During emergency page the station will queue all incoming calls. When emergency page ends, the station will accept calls again.
    • If a station sends a busy override to a station listening on emergency page, then emergency page will stop
  • If a station is in a regular point-to-point call when receiving a multicast paging of defined priority:
    • LOW / NORMAL priority: The paging audio is not played
    • HIGH priority: The paging audio is mixed with the normal call
    • EMERGENCY priority: The regular call is muted

Configuration of Multicast streaming sources

Stream from FFMPEG

Stream radio source in g722:

ffmpeg -re -i "http://lyd.nrk.no/nrk_radio_p13_mp3_h" -filter_complex aresample=16000,asetnsamples=n=160 -ac 1 -acodec adpcm_g722 -f rtp rtp://239.1.20.5:61111



Stream radio source in g711u/pcmu:

ffmpeg -re -i "http://lyd.nrk.no/nrk_radio_p13_mp3_h" -filter_complex aresample=8000,asetnsamples=n=160 -ac 1 -acodec pcm_mulaw -f rtp rtp://239.1.20.4:61110


Stream from VLC

To stream from VLC does not work due to variable payload size, same problem as on SNOM (Not possible to fix this). http://wiki.snom.com/Category:HowTo:Multicast_Audio#Multicast_via_VLC


Listen to Edge Group Call

Go to Edge Controller -> Group Calls and use the multicast address/port written in each group call. Note that the clients making the group call must use same codec as chosen for multicast paging.


Listen to Pulse Group Call

Go to Pulse Server -> Group Call and use the multicast address/port written in each group call. Note that the clients making the group call must use same codec as chosen for multicast paging.


Use Multicast Paging with Asterisk

Call 8998 to speak on multicast address:

exten => 8998,1,Dial(MulticastRTP/basic/239.1.2.4:61101//c/g722))


Use Multicast Paging with Freeswitch iPBX

<extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243$"> <action application="answer"/> <action application="esf_page_group" data="239.1.20.5 61082"/> <action application="esf_multicast_write_codec" data="G722"/> </condition> </extension>


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