Cees de Groot wrote:
Nope, but it does increase the chance that you can't get a voice stream setup in both directions. Our tests have shown that keeping Asterisk in the loop in a heterogeneous environment (with more combinations of SIP UA's, proxies/firewalls with various combo's of NAT and SIP support, etcetera than I care to list here) is quite a bit more reliable. That's
Sure, you must not enable re-INVITE for peers that you don't know for a _fact_ will support it properly. Anything behind a NAT/firewall will be prone to problems, and proxies that don't handle re-INVITE properly will be problematic as well.
However, if you don't turn it on unless you are sure that device can handle it, it works very well and is quite useful. We use it between our network of Asterisk servers so that the media takes the shortest path possible at all times.
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