On Feb 28, 2007, at 1:01 AM, Helmut Kuper wrote:
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Hi,
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 sip accounts to do outbound calls into a external VoIP-Cloud/domain. My picture of it is as follows:
1. uac1@mydomain.com makes an outbound call to uac2@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 call as a member of externaldomain.com eg. freeswitch@externaldomain.com.
4. freeswitch bridges the RTP stream between uac1 and freestwitch's own call.
This means uac1 is only talking to freewitch while freeswitch is talking 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 "Invalid 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 which 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. /b