[Freeswitch-users] Can Freeswitch and OpenSER Co-reside on Same Server?

[Kristian Kielhofner]

On Fri, Feb 5, 2010 at 1:36 PM, Henry Huang <red.rain.seven at gmail.com> wrote:
> Kristian:
>
> Can you point me to the wiki link where it describes how to do the header
> rewriting and number formatting and topology hiding?
> I am also looking into OpenSIPS to be a session boarder controller, but if
> freeswitch is already able to do all these, then I think it's easier for me
> to stick with it since I already learn and do a lot with it already.
>
> Thanks,
>

Henry,

  By default (when using bridge) FreeSWITCH will generate a new leg
with a fresh set of headers.  Various channel variables can be used to
influence the values of some of these:

Request URI: based on bridge string (411 at sip.provider.com) - can also
include transport, port, uri params, etc
To: same as Request URI (possible minus some of the params)
From: Determined by gateway config (if used: use-callerid-in-from) and
effective_caller_id_number/effective_caller_id_name
RPID/PAI: sip_cid_type

  The other params will be taken from the Sofia config (session
timers, etc).  Use multiple profiles for internal and external
networks (just like in the samples).

  FreeSWITCH does topology hiding by default - as a B2BUA (regardless
of "mode") it will do topology hiding by creating a new channel/leg.
None of the IP addresses etc, from the original channel are visible.
Note that this doesn't count the SDP if you are using bypass media or
proxy media.  If you want *full* SBC style topology hiding with media
you can't use these modes but you'll pay for it in performance.

  Unless you use the uac/uas modules and/or some textops based
manipulation in OpenSER, all OpenSER can do is copy most of the
headers/body, add Record-Route and Via headers, and forward the
message to the next hop while leaving all IP addresses, etc intact.
OpenSER/OpenSIPS can be a fairly general purpose SIP server but it's
main function is a fairly strict RFC 3261 compliant proxy.

  I should also point out that OpenSIPS does have a B2BUA module.  I
myself would much rather just use FreeSWITCH.  You know - the best
tool for the job.  They're both EXCELLENT pieces of software and
between the two of them you can build incredible VoIP solutions and
networks.

  I use FreeSWITCH to interface with each of my carriers.  FreeSWITCH
runs on the same machine as our main SIP proxy running in
bypass_media.  The SIP proxy (OpenSER) handles all of the requests for
our servers/customers (servers we provision and control) while
FreeSWITCH (usually in bypass media) interfaces with each of our
carriers to make everyone happy at the signaling level.  I have a
profile for our network and a profile for each carrier.  No need to
worry about different caller id formats, number formats (e.164),
transports, caller id, etc.  On one machine FreeSWITCH regularly does
over 1200 channels using about %20 CPU.  It's a four year old Dell
1850 :).

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com