Aaron Daniel wrote:
I know it's not a NAT environment, but the way we got around that was by setting nat=yes in the sip.conf. nat=yes basically just tells the server to stick around during the conversation so you don't lose the rtp stream.
No, it doesn't. That may be a side effect of it, but that is not the purpose of that setting at all. Asterisk can easily stay in the RTP stream even for nat=no endpoints, if it needs to.
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