[sip-comm-dev] Can I run mutilple sip clients in a user PC


I'm actually having a similar problem. I'm running Sip-Communicator on a Sun Ray Server
(Sun's SRSS v3.1 b21) where potentially many users run all with the same IP address, just like SSH or Shell
sessions. In order for the users not to collide with each other they need to listen to
different SIP/RTP ports for each user and the OS has to provide the security on who can access what.
This same issue also comes up on Windows Terminal Server users, (they all run at the server's IP as well)
now that Microsoft's latest protocol update supports bi-directional audio/video connections from WTS servers
to WTS clients.
Can this "always use the same RTP ports" be made a configurable option? Then those of us who require
"user specific, system allocated, non-privileged, non-root, ports (1024+)" could enable it?
Best regards,


Date: Tue, 5 Jul 2005 17:26:32 +0100 (BST)

From: SubashSubbiah <suppa_s@yahoo.co.in>

Content-Type: text/plain; charset=iso-8859-1
Subject: [sip-comm-dev] Can i run mutilple sip clients in a user PC


      I tried to run three sip-communicator in a single PC. Then I monitered the packets using
Ethereal. In RTP pckts the source and destination Port is the same.

This was actually done on purpose as it helps in NAT & Firewall traversal.

I have configured different
ports before starting. Kindly get back to me whether it is possible to RUN multiple SIP in a PC.

Well it depends. It should be possible but only one of the clients would
have access to capture devices. You could try and configure the other
one to read and send the contents of a file (i.e. using the MEDIA_SOURCE
property )



Can u please eloborate on how this solves firewall issues. Let me try to explain my problem

I have two clients A and B running on same (or different) machines. Lets say "A" uses port 5004
and "B" uses port "6004". The following is the expected behavior.

1. "A" invites "B", the SIP request will have the audio port as 5004.
2. "B" receives this and knows that "A" will be send AND receive the data at port 5004.
3. "B" sends an OK with audio port 6004. Now, A knows that B will send AND receive in port 6004.
4. Hence, all RTP data from A to B must have src port as 5004 and dst port as 6004...
5. All data from B to A must have src port as 6004 and dst port as 5004.

The problem that I face is, instead of step 4, A sends data with SRC PORT AS 5004 and DST port as
5004 and 6004. Similarily from B to A. This I beleive must be an issue in SIP communicator. Is my
understanding right.

the boy next door,

evr ur's


Luv & b Luvable

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