[Last Call] Learn about multicloud storage options and how to improve your company's cloud strategy. Register Now

x
?
Solved

SIP Confrence Call hold?

Posted on 2004-10-05
4
Medium Priority
?
577 Views
Last Modified: 2010-08-05
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?

                         
0
Comment
Question by:neron
[X]
Welcome to Experts Exchange

Add your voice to the tech community where 5M+ people just like you are talking about what matters.

  • Help others & share knowledge
  • Earn cash & points
  • Learn & ask questions
  • 2
  • 2
4 Comments
 
LVL 12

Expert Comment

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?
0
 

Author Comment

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??
0
 
LVL 12

Accepted Solution

by:
Joel_Sisko earned 500 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
0
 

Author Comment

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



//neron
0

Featured Post

Learn Veeam advantages over legacy backup

Every day, more and more legacy backup customers switch to Veeam. Technologies designed for the client-server era cannot restore any IT service running in the hybrid cloud within seconds. Learn top Veeam advantages over legacy backup and get Veeam for the price of your renewal

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

In my office we had 10 Cisco 7940G IP phones that were useless as they were showing PROTOCOL APPLICATION INVALID when started. I searched through Google and worked for a week continuously on those phones, and finally got them working. This is a di…
Why do some people recommend buying business VoIP from an ISP? What are the benefits to my company? What are the costs?
In this video, Percona Director of Solution Engineering Jon Tobin discusses the function and features of Percona Server for MongoDB. How Percona can help Percona can help you determine if Percona Server for MongoDB is the right solution for …
Please read the paragraph below before following the instructions in the video — there are important caveats in the paragraph that I did not mention in the video. If your PaperPort 12 or PaperPort 14 is failing to start, or crashing, or hanging, …
Suggested Courses

650 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question