[asterisk-users] Does SIP limit to 3-way conference?

[Kevin P. Fleming]

On 07/23/2010 01:12 AM, Karl Fife wrote:
>> Cassius Smith wrote:
>>> Hello all,
>>> I'm in final testing stages and preparing training for a new Asterisk
>>> rollout. I'm replacing a Cisco Call Manager system, and re-flashing
>>> the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call,
>>> I use the "Confrn" softkey to invite other participants. I can add one
>>> other participant
>>
>> I think the phone itself is limited to 3 calls per line.
>>
>> Doug
> 
> It's got nothing to do with SIP.  The phone itself is doing the audio 
> mixing, and the phone is limiting you to 2 call legs (plus you, making a 
> 3-way call).  You don't usually see phones that will bridge more than 2 call 
> legs because the required technical resources generally exceed those of the 
> endpoint.
> 
> Some 'better' phones such as the Polycom Soundpoint IP 6xx series phones 
> will mix as many as 3 call legs (plus you, making a 4-way call).  Still, the 
> phone itself is bearing the burden of the audio mixing which explains the 
> limitation.
> 
> To conference more than 4 parties, you need a conference bridge (such as 
> MeetMe). However, you can get clever with Asterisk and allow your end-user 
> to transfer calls directly to a MeetMe conference with a single keypress! 
> You can enable your end-user to collect conference participants one-by-one, 
> rather than burdening your called parties with the need to dial in to a 
> special phone number with a special access code!

And the real reason the OP is noticing this change is that their phones
were previously using Call Manager (SCCP) firmware, where all the
conferencing is done on the server. In that scenario, there is no
3-party limit for phone-managed conferences, because the phone is
literally only managing the conference, not doing the audio
decoding/mixing/encoding.

There aren't any commonly-deployed SIP methods of managing server-hosted
conferences, although there are some being worked on (XCON in the IETF,
for example), so right now with SIP phones you are limited to the number
of channels the phone can mix itself if the phone is managing the
conference.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org