Multicast Paging (SIP)
From Zenitel Wiki
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.
To configure a station to receive Multicast Paging, log into the IP Station and select SIP Configuration > 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. 188.8.131.52. Multicast IP addresses are in the range from 184.108.40.206 to 220.127.116.11.
- 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.
- 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.
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 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
Stream radio source in g722:
ffmpeg -re -i "http://nrkradio-f.akamaihd.net/i/ostlandssendingen_0@4273/index_64_a-b.m3u8?sd=10&dw=31&rebase=on" -filter_complex aresample=16000,asetnsamples=n=160 -ac 1 -acodec adpcm_g722 -f rtp rtp://18.104.22.168:61111
Stream radio source in g711u/pcmu:
ffmpeg -re -i "http://nrkradio-f.akamaihd.net/i/ostlandssendingen_0@4273/index_64_a-b.m3u8?sd=10&dw=31&rebase=on" -filter_complex aresample=8000,asetnsamples=n=160 -ac 1 -acodec pcm_mulaw -f rtp rtp://22.214.171.124:61110
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 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/126.96.36.199:61101//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="188.8.131.52 61082"/> <action application="esf_multicast_write_codec" data="G722"/> </condition> </extension>