[SR-Users] Help on setting UP SIP PRoxy

[Mack Hendricks]

Hey Ayub,

I would take it one step at a time.  Kamailio is a toolkit with a bunch of modules.  I find it useful to first figure out what features you need in a proxy.   This includes understanding what are you trying to protect your backend media servers from, problems you are currently having and/or how are you trying to scale your backend servers.  

Assuming you are using Asterisk or FreeSwitch as your backend media servers, here are some questions that might help derive the features you want:

- Are the backend servers being overwhelmed by REGISTER requests?
- Are you providing the proxy for phone end points, PBX’s or both
- What’s your High Availability Requirements?  Are the backend media services replicated so it doesn’t mater what server gets the original INVITE requests?
- Are you handing registrations for 1 SIP domain or multiple SIP domains?
- Do you want to maintain CDR’s at the proxy or leave it to the backend media servers

Mack Hendricks / Head of Support / dOpenSource
web: http://dopensource.com <http://dopensource.com/>
support: +888-907-2085
dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services


> On Mar 26, 2018, at 1:26 AM, Mohammed Ayub <payubpashap at gmail.com> wrote:
> 
> HI,
> 
> Myself Ayub from Phonology, Bangalore, India.
> 
> We are trying to set up an SIP proxy server using Kamailio with Siremis as web interface.
> 
> Have installed and configured kamailio and siremis but need help in config if sip domain , users  and etc...
> 
> Please revert.
> 
> Regards
> Ayub
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