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SIP Confrence Call hold?

Posted on 2004-10-05
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Hi,

Is it possible to put a SIP confrence call on hold and make a new call and later pick up the confrence call again? Can this be done using the UPDATE Session Description a=inactive or does that only work between 2 users.

Heres my case:
Confrence                                  User A                            User B
     | <--------Conversation---------> |                                    |
     |                                              |                                    |
     | <--------UPDATE inactive------  |                                    |
     | --------------200 OK -----------> |                                    |
     |  =====   On hold ======  |                                    |
     |                                              |=====New call===== |
     | <--------UPDATE sendrecv------|                                     |
     | -----------200 OK --------------->|                                    |
     |  <-----starts talking again-----> |                                    |

Is this possible?

                         
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Question by:neron
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by:Joel_Sisko
ID: 12331557
Before I can answer, what version of the SIP Protocol are you using? Also, is this a custom devlopment app or some other thirs party application you need to modify?
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by:neron
ID: 12336755
Hi Joel,

I'm using SIP v2.0. The user agent is a JAIN-SIP based app and the proxy/redirect I don't know yet. Perhaps the only way to get this working is by developing a custom proxy/redirect aswell??
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Joel_Sisko earned 125 total points
ID: 12352536
First, in regarding the UPDATE request, the intent of this particular request is really meant for updating the parameters (codec, route, etc) of a session (single or multicast) in the early stages of the initial INVITE request. Section 5 of RFC3311 goes over this in more detail. It is recommended that a Re-INVITE request is used instead to update the parameters of a given session once it has been confirmed (existing).

In regards to conferencing, the ability to enter in an out is dependent upon the architecture/authentication being used . If you  use a simple end mixing architecture, utilizing UA’s to establish and mix the audio streams, then you may have problems putting the conference on hold, calling a separate party, terminating the call and re-entering the original conference. In standard TDM practice for legacy PBX’s, if the originator of a conference call leaves the conference, the entire conference call is terminated.

Now by default, what you are looking to do, is really the nature of conferencing, meaning that in order to add a party to a conference, you normally put the current participants on hold and establish your next call, once the additional call is establish, you then invite them into the conference. In your case you will establish the call but never invite the participant into your conference. Now, possible issues might arise when you terminate the independent call or have a time out issue if you do not rejoin the existing conference within a set timeframe.

Now if you go with server architecture, it just about alleviates any and all of theses issues. You can establish an ad-hoc or meet-me conference. This allows any and all participants to join in and out of the conference as needed. The server handles the mixing and distribution of the audio streams, allowing flexibility for the UA’s in the conference.

You might want to try to utilize some open source products such as

SER which can be found at,  http://www.iptel.org/ser/  or

Asterisk PBX with the Meet-me module,   http://www.asterisk.org/

Also if you are creating a  custom application development, might I recommend a tool to help out in seeing what is going on with all the requests:

SIP Scenario    http://www.iptel.org/~sipsc/

Kindest regards
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by:neron
ID: 12355291
Great answer! Thank's a lot!!!



//neron
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