View source for Multicast Paging (Edge/SIP)

{{ES}}
'''Multicast Paging''' enables Zenitel devices in '''Edge''' or '''SIP''' mode to receive RTP multicast streaming from e.g. 3rd party iPBX (e.g. Asterisk). Up to 10 different Multicast paging groups can be defined.
{{note|
* ''Required software: [[:Category:Stations#Turbine_stations|Turbine]] 5.0 or newer. [[:Category:Stations#INCA_stations|INCA]] 2.10 or newer'' }}

== Configuration ==
To configure a device to receive Multicast Paging, [[Logging into an IP Station|log into the device]], enable '''Advanced Configuration''' mode, and select '''SIP/Edge Configuration''' > '''Multicast Paging''':

<gallery caption="Multicast Paging configuration" widths="650px" heights="420px" perrow="2">
File:MulticastPaging SIP ENA.PNG|''Configuration page in ENA Amplifiers and for Zenitel devices running firmware 8.2 and newer''
File:MulticastPaging SIP.PNG|''Configuration page in Zenitel devices running firmware 8.1 and older''
</gallery>

* '''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 port number must be within the port range 61020-61250, recommended range is 61080-61250. Using odd number for port is not recommended. 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|'''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 (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.
** 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 page 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 ===
[https://ffmpeg.org/ FFmpeg] is the leading multimedia framework, able to decode, encode, transcode, mux, demux, stream, filter and play pretty much anything that humans and machines have created.


Stream radio source in g722:
{{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://239.1.20.5:61080</nowiki>}}

  
Stream radio source in g711u/pcmu:
{{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://239.1.20.4:61082</nowiki>}}


{{note|If multiple ethernet interfaces are in use on the computer running ffmpeg, the multicast is sent to the interface with the lowest metrics (highest priority). Disable unused interfaces, or make sure that the interface used for streaming has the lowest metrics.}}

=== 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

<br>

=== Listen to Edge Group Call===
Go to [[Group_Call_(Edge)#Configuration_of_group_calls|Edge Controller -> 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.

<br>

=== Use Multicast Paging with Asterisk===
Call 8998 to speak on multicast address:
{{Code|exten <nowiki>=</nowiki>> 8998,1,Dial(MulticastRTP/basic/239.1.2.4:61084//c/g722))}}

=== Use Multicast Paging with Freeswitch iPBX=== 

{{Code|<nowiki><extension name="rtp_multicast_page">
  <condition field="destination_number" expression="^pagegroup$|^7243$">
    &#60;action application="answer"/&#62;
    &#60;action application="esf_page_group" data="239.1.20.5 61086"/&#62;
    &#60;action application="esf_multicast_write_codec" data="G722"/&#62;
  </condition>
 </extension></nowiki>}}

[[Category: SIP intercom - Configuration]]
[[Category: IC-EDGE Configuration]]
[[Category: Audio Settings and Audio Features]]