[asterisk-dev] [Code Review] RTP monitoring branch: initial review

Kevin P. Fleming kpfleming at digium.com
Thu Oct 8 07:50:01 CDT 2009

Tzafrir Cohen wrote:

> 2. Even if there were someone who responded, maintaining a proper
> session would be a waste of time and resources. I need to start pushing
> recorded data ASAP.

Also, it would require *two* SIP sessions, since there are two audio
streams being sent to the recording server for a single call.

Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org

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