[Asterisk-Users] P2P RTP without SIP re-invites

David Luyens d at vt4.net
Mon Feb 2 23:38:39 MST 2004


Hi Adam, could you share your clustering setup?

David

-----Oorspronkelijk bericht-----
Van: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] Namens Low, Adam
Verzonden: maandag 2 februari 2004 12:11
Aan: 'asterisk-users at lists.digium.com'
Onderwerp: RE: [Asterisk-Users] P2P RTP without SIP re-invites


Apologies for the belated reply but I've spent the weekend fighting DDoS
attacks against Superbowl sites ... )c;

Ok, well I am not sure what went wrong with previous testing but I have
tried this again with Cisco 7940's and Cisco AS5300's and indeed the RTP
stream flows directly between end-points retaining SIP signalling via
Asterisk. This is exactly the operation I had hoped for. I had
previously tested with my home 7940 which it behind NAT without success
and so will re-test this this evening.

Thanks for all the responses and related discussion on clustering
Asterisk, thanks to those I now have a running cluster of 3  Asterisk
servers each with mirrored sip.conf and extensions.conf built
dynamically from a MySQL backend database.

Rgds, Adam

-----Original Message-----
From: Brancaleoni Matteo [mailto:mbrancaleoni at espia.it]
Sent: 31 January 2004 13:20
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] P2P RTP without SIP re-invites


hi
> 
> I guess this would work if both Alice and Bob were NAT'ed on the 
> inside of the same NAT box. The problem is that if Alice and Bob both 
> have NAT=yes and CANREINVITE=yes and they're on separate NAT'ed 
> networks, the call is broken. So it's a dangerous configuration.

nope. I have a public * server (beta server for a free VoIP service), on
a public IP. and some sip phones around , like one in my home, behind
nat, one in my office (another nat) and some others at my coworkers
home... all behind nat. and are different nat box, do you agree? that
works ok, I have RTP passing directly from one endpoint to the other...
no RTP on the public * server. No stun is used. The phones are
budgetones in this case. All are configured with nat=yes on asterisk
side. or I missing something?
-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


********* DISCLAIMER ********* 

This message and any attachment are confidential and may be privileged
or otherwise protected from disclosure and may include proprietary
information. If you are not the intended recipient, please telephone or
email the sender and delete this message and any attachment from your
system. If you are not the intended recipient you must not copy this
message or attachment or disclose the contents to any other person 


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list