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


#1

Hello Subash,

I don't really see a problem with your session. The 2 clients in the session - have IPs 59 and 243, and they use a proxy with the address 210. In its SDP offer the 59 client announces port 18010. Client 243 announces port 18014. What happens from that point on is perfectly in order. 59 sends data to port 18014 and 243 sends data to 18010. Indeed they are sending data from the same that their packets are destined to but that does not break any rule whatsoever. What's more it helps NAT and firewall traversal as some NAT boxes would let client A to send packets to client B on port X only if B has previously sent packets (to A) FROM that port X.

2. "B" receives this and knows that "A" will be send AND receive the data at port 5004.

Actually "B" doesn't have the right to know anything about the port that A is going to use to send data. A is free to chose any port for sending its data.

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.

I don't see why this should be the case. As I said - there are no "MUST"s concerning the port that a user agent uses to send media. Both connection address and port in SDP offers and answers indicate a place where the US would like to _receive_ data.

Hope this helps
Emil

SubashSubbiah wrote:

···

Hi Emil Ivov,
            Thanks for your quick reply.
Here with i have attached the Ethereal captured packet.
Kindly update me with yoour findings.

Note : you can note that the Source & Destination port in the User Datagram Protocal for RTCP/RTP packets were same.

Thanks & Regards,
S.Subash.

--- Emil Ivov <emil.ivov@gmail.com> wrote:

May I please see a saved ethereal dump of the session that you are talking about? I had originally misunderstood the problem, sorry. I might have a clue of what's going on but I'll need to see the dump

Cheers
Emil

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

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. Similarily from B to A. This I beleive must be an issue in SIP communicator. Is my
understanding right.

Thanks for your efforts,
S.Subash.

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