[Freeswitch-users] Freeswitch as b2bua
brian.west at mac.com
Wed Feb 28 01:31:42 PST 2007
On Feb 28, 2007, at 1:01 AM, Helmut Kuper wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> I want to know whether freeSWITCH is able to be a b2bua proxy for both
> sip and rtp or not. sofia has a registration section to Register
> freeswitch to one or more upstream VoIP provider. I understand this
> feature in that way, that UAC registered to freeswitch can use this
> accounts to do outbound calls into a external VoIP-Cloud/domain. My
> picture of it is as follows:
> 1. uac1 at mydomain.com makes an outbound call to uac2 at externaldomain.com
> 2. freeswitch gets this call, answered it and sets up a RTP stream
> to uac1.
> 3. freeswitch uses the destination address of uac1 and establish a
> as a member of externaldomain.com eg. freeswitch at externaldomain.com.
> 4. freeswitch bridges the RTP stream between uac1 and freestwitch's
> This means uac1 is only talking to freewitch while freeswitch is
> to uac1 destination. So freeswitch acts like a man in the middle.
> Am I wrong ?
This is how it should work but that totally depends on config. can
you provide that also?
> In my test environment freeswitch registers to the upstream provider,
> but when I start a call from uac1 to a uac in that upstream provider
> domain. freeswitch takes it and send an invite to upstream. In this
> invite message I can find a source sip address which is that from uac1
> instead of freeswitch's registration account for upstream domain. The
> upstream sip proxy sends first a Trying and then a 488 message
> Session Description". The 488 message contains a Warning named "
> Warning: 301 xxx.xxx.xxx.xxx 'invalid transport IP address'"
Can you provide log output so we can see what is going on.
> I think this reffers to the forwarded uac1 source sip address, so
> that I
> think that the upstream sip proxy doesn't accept source addresses
> doesn't belog to that upstream sip domain.
> Any ideas how I can convince freeswitch to act as a UAC-Proxy for rtp
> and sip?
It already does what you want where its totally in the middle.
More information about the FreeSWITCH-users