4 Feb
2014
4 Feb
'14
10:56 a.m.
If I want to set up a SIP redirector for small call volumes (eg. <1/second), can Freeswitch do as good a job as Kamailio? The idea is that we get a call in from a PSTN-SIP gateway and then re-route to another box depending on the number being called. Just wondering if there's any difference in the signalling, which I presume is either a 302 or a SIP re-invite (but please correct me if I'm wrong!). At what kind of call volumes might one choose to switch to a more dedicated SIP proxy?