[FAQ] What is the maximum amount of participants in a conference call?

Polycom Employee & Community Manager

[FAQ] What is the maximum amount of participants in a conference call?


Local Conferencing


The phone can conference together the local user with the remote parties of a certain number of independent calls by using the phone’s local audio processing resources for the audio bridging.


Polycom Phones utilizing Microsoft Skype for Business are utilizing the S4B server for this.


There is no dependency on network signaling for local conferences.


Phone Model Type of Conference supported nWayConference
VVXphones  Support three-way calls Yes
SoundStructure VoIP card Support three-way calls No
SoundStation IP 4000 Support three-way calls No
SoundStation IP 5000 Support three-way calls No
SoundStation IP 6000 Support three-way calls No
SoundStation IP 7000 Support three-way calls No
SoundStation Duo Support three-way calls No
SoundPoint IP 450, 550, 560, 650, 670 Support four-way calls Yes
SoundPoint IP 321, 331, 335 Support three-way calls No
RealPresence Trio 8800 (SIP) Support five-way calls Yes



In order to utilize more than 3 participants with the compatible phones you will require the feature.nWayConference.enabled="1" parameter.


A compatible phone will display the manage Option




And individual Participants can be removed or set to mute etc. 




Centralized Conferencing


Centralized conferences use an external audio bridge or a compatible SIP Server, available via a central server, to create a centralized conference call.


The Parameter voIpProt.SIP.conference.address is used to specify a destination.


For the example voIpProt.SIP.conference.address="5002@" the phone would initially send a SIP INVITE to this central server.




Phone Logs:

0810133536|sip  |2|00|CStkCall::Conference new call 0x188d0bc sending INVITE to conf URI '5002@'
0810133536|sip  |0|00|    INVITE sip:5002@ SIP/2.0
0810133536|sip  |0|00|    Via: SIP/2.0/UDP;rport;branch=z9hG4bKc79f7fb2CD77325D
0810133536|sip  |0|00|    From: "3031" <sip:3031@>;tag=2942A73E-4061CA19
0810133536|sip  |0|00|    To: <sip:5002@>
0810133536|sip  |0|00|    CSeq: 2 INVITE
0810133536|sip  |0|00|    Call-ID: a1db769c83b223379946436ffa5b572d
0810133536|sip  |0|00|    Contact: <sip:3031@>
0810133536|sip  |0|00|    User-Agent: Polycom/ PolycomVVX-VVX_600-UA/
0810133536|sip  |0|00|    Accept-Language: en
0810133536|sip  |0|00|    Supported: replaces,100rel
0810133536|sip  |0|00|    Allow-Events: conference,talk,hold
0810133536|sip  |0|00|    Authorization: Digest username="3031", realm="asterisk", nonce="447e4919", uri="sip:5002@", response="c7b384ea1b5d44843c2669b8ac8dd861", algorithm=MD5
0810133536|sip  |0|00|    Max-Forwards: 70
0810133536|sip  |0|00|    Content-Type: application/sdp
0810133536|sip  |0|00|    Content-Length: 508
0810133536|sip  |0|00|    
0810133536|sip  |0|00|    v=0
0810133536|sip  |0|00|    o=- 1502368536 1502368536 IN IP4
0810133536|sip  |0|00|    s=Polycom IP Phone
0810133536|sip  |0|00|    c=IN IP4
0810133536|sip  |0|00|    b=AS:512
0810133536|sip  |0|00|    t=0 0
0810133536|sip  |0|00|    a=sendrecv
0810133536|sip  |0|00|    m=audio 2242 RTP/AVP 9 102 0 8 18 127
0810133536|sip  |0|00|    a=rtpmap:9 G722/8000
0810133536|sip  |0|00|    a=rtpmap:102 G7221/16000
0810133536|sip  |0|00|    a=fmtp:102 bitrate=32000
0810133536|sip  |0|00|    a=rtpmap:0 PCMU/8000
0810133536|sip  |0|00|    a=rtpmap:8 PCMA/8000
0810133536|sip  |0|00|    a=rtpmap:18 G729/8000
0810133536|sip  |0|00|    a=fmtp:18 annexb=no
0810133536|sip  |0|00|    a=rtpmap:127 telephone-event/8000
0810133536|sip  |0|00|    m=video 2244 RTP/AVP 109 34
0810133536|sip  |0|00|    a=rtpmap:109 H264/90000
0810133536|sip  |0|00|    a=fmtp:109 profile-level-id=42800d
0810133536|sip  |0|00|    a=rtpmap:34 H263/90000
0810133536|sip  |0|00|    a=fmtp:34 CIF=1;QCIF=1;SQCIF=1


Subsequent calls adding the conferenced party would then use the SIP REFER method to be placed into the conference.




0810133536|sip  |0|00|    REFER sip:3051@ SIP/2.0
0810133536|sip  |0|00|    Via: SIP/2.0/UDP;rport;branch=z9hG4bK7837f44881821A4B
0810133536|sip  |0|00|    From: "3031" <sip:3031@>;tag=A77D1E46-83CDA941
0810133536|sip  |0|00|    To: <sip:3051@;user=phone>;tag=as159a99e6
0810133536|sip  |0|00|    CSeq: 4 REFER
0810133536|sip  |0|00|    Call-ID: 499db673c842b2a74a7514c42d5b572d
0810133536|sip  |0|00|    Contact: <sip:3031@>
0810133536|sip  |0|00|    User-Agent: Polycom/ PolycomVVX-VVX_600-UA/
0810133536|sip  |0|00|    Accept-Language: en
0810133536|sip  |0|00|    Refer-To: <sip:5002@>
0810133536|sip  |0|00|    Referred-By: <sip:3031@>
0810133536|sip  |0|00|    Authorization: Digest username="3031", realm="asterisk", nonce="24781e88", uri="sip:3051@;user=phone", response="d7d4dd59b095a44e255ff631407163cd", algorithm=MD5
0810133536|sip  |0|00|    Max-Forwards: 70
0810133536|sip  |0|00|    Content-Length: 0

The above scenario the Polycom Phone 3031 conferenced the Phone 3051 into the Meetme conference 5002.


Example from the Asterisk meetme.conf:


conf => 101,123456
conf => 102


Example from the Asterisk extensions.conf:



exten => 5001,1,MeetMe(101,123456) exten => 5002,1,MeetMe(102) ...


Above examples would either use:


  • voIpProt.SIP.conference.address="5001@xxx.xxx.xxx.xxx" to utilize 5001 and a PIN number of 12345
  • voIpProt.SIP.conference.address="5002@xxx.xxx.xxx.xxx" to utilize 5002 and no PIN number.


Hosting Conference Calls


You can create a conference with up to two other parties (or more pending above table).


After you set up a conference, you can place the conference call on hold, split the conference call into two calls on hold, or end the conference call (and your connection to the conference call participants).


Setting Up Conferences


You can set up a conference in one of two ways:


  • Using the Confrnc soft key.
  • Using the Join soft key, if you have an active call and a call on hold.

To set up a conference using the Conference soft key:

  1. Call the first party.
  2. Press the Confrnc soft key. The active call is placed on hold.
  3. Enter the number of the second party, and then press the Send soft key.
  4. When the second party answers, press the Confrnc soft key to join all parties in the conference, as shown next.
Please be aware:

The purpose of these forums is to allow community members collaborate and help each other.
Questions posted here do not follow Polycom’s SLA guidelines.
If you require assistance from Polycom technical support, please open a
web service request or call us .

The above is necessary in order to track issue internally within Polycom.

You are welcome to post more questions or configuration or logs for other community members to look at but if your issue requires a fix via Polycom you must go via the official support structure.

Please ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

This forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Polycom employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and may be answered on weekends, bank holidays or personal holidays.