On Fri, Feb 29, 2008 at 4:03 AM, Ron <[EMAIL PROTECTED]> wrote: > Hi Greyman, > > Should it look like this now? Can i use 2 SIP Proxies just to make sure > i have a backup? will that cause any problem again with regards to > calling extension to extension? Extensions will register on the asterisk > still? How about outbound calls to other SIP provider or a telcobridge, > SIP proxy will handle that also? Basically asterisk will ask SIP proxy > of everything? Will RTP stream still go thru asterisk? > > Also, i plan on setting these up as a Virtual PBX for multiple offices, > which means company A can only use Trunks for A, B is for Trunk B etc > etc. Does outbound to trunks have any issues? or problem is just > basically calling extension to extension? > > > [other voip provider] [telcobridge] -- [pstn] > | | > -------------------------------------------------------------------- > [ SIP Proxy ] > -------------------------------------------------------------------- > > | | | | > [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] > | | | | > -------------------------------------------------------------------- > | mysql cluster | > -------------------------------------------------------------------- > > > Thanks > > Regards, > Ron
Hi Ron, Yep it starts to get tricky :). There will be slight difference depending exactly on what you need to accomplish. I work for a VoIP Proivder that provides services to users in internet land so our set up is designed for that. If you've got VPNs or are on a LAN things will be different. Two SIP Proxy's are definitely a good idea, you can load balance your users across them using DNS SRV records, DNS Round Robin, IP Load Balancer (although then you prob should have two load balancers). If you're just starting your build network build or only have < 1000 users the extra SIP Proxy should go to the bottom of the list. SIP Proxy's such as OpenSER are pretty stable and don't do anywhere near as much work as the media server. It's the fault tolerance on your Asterisk servers that is the most critical thing. They do a lot more work and in my experience with them (4+ years) they are a lot more likely to crash than your SIP Proxy. With two SIP Proxy's you have an additional problem in that now you need to set the outboundproxy field in the Asterisk realtime database to the value of the proxy the user agent came through. Asterisk can't do that for you (as far as I know) so you could possibly use the SIP Proxy to do registrations or use a custom SIP Registrar. Both are a good idea as they take registration load away from Asterisk and this can be VERY significant as your user base grows. We use a custom SIP Registrar. For outbound trunking we go directly from Asterisk to the terminating gateway no SIP Proxy involved. For inbound trunking we do go through the SIP Proxy for the same reason you get users to. Incoming calls are going to be more reliable if they are not tied to a single Asterisk server (I guess you could use SRV records for your Asterisk servers for inbound trunking as well but then you're kind of duplicating the role of the SIP proxy). The RTP stream will always be between the users and Asterisk the SIP Proxy is never invovled. If you send an RTP packet to a SIP Proxy and it will just shake its head in an irritated manner and ignore you. Regards, Greyman. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
