On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer. > Have you tried > rtp_symmetric to see if the endpoint sends audio to asterisk if > asterisk > can send audio back to the client? That would require using chan_pjsip wouldn't it? Not that I am opposed to trying that. I need to use chan_pjsip at some point to be able to authenticate to my SIP provider for SIP SIMPLE anyway. But will rtp_symmetric really solve the problem? Isn't the problem the setting up of the RTP session, so there is no address and port that it receives from yet? > Alternatively if your SIP Proxy is also a Media proxy you could set > the > media_address on the endpoint to be your proxy and let your proxy > handle > proxying the media to your endpoint. The idea of sending my media out of the LAN (where I have almost zero latency) and introducing the latency of a round trip to the proxy and back doesn't seem like a great solution. Cheers, b.
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