<div dir="ltr"><div><div>Hi David,<br>I got you, but FS can't accept multiple m=audio lines, which makes it incompatible for SIPREC.<br></div>I don't think there is plans to modify the SIP diver in freeswitch, in order to <br>support this protocol. Enabling additional RTP engines for a single SDP session is not trivial.<br><br></div><div>Regards,<br><br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jun 8, 2016 at 10:03 PM, David Villasmil <span dir="ltr"><<a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello Miguel,<div><br></div><div>Thanks for your reply, but I think maybe I didn't explain myself correctly, let me try again:</div><div><br></div><div>A recording server receives INVITEs including multiple "m=" which will be used as rtp source so the server can record an ongoing conversation happening somewhere else. As per RFC4566 an SDP may have as many m= as it wants (<a href="https://tools.ietf.org/html/rfc4566#page-22" target="_blank">https://tools.ietf.org/html/rfc4566#page-22</a>).</div><div><br></div><div>The idea is to use SIPREC the following way:</div><div><br></div><div>The voice server will choose the call to record and send a 3rd leg to the recording server, which will receive the INVITE ans answer with 200 OK and start recording the rtps.</div><div><br></div><div>Regards,</div><div><br></div><div>David</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jun 8, 2016 at 12:40 PM, Miguel Oyarzo <span dir="ltr"><<a href="mailto:miguelaustro@gmail.com" target="_blank">miguelaustro@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><br><br></div>Nighter FS nor Asterisk will process two same-type RTP streams in the same SDP. The 2nd RTP streams will be declined and answered with a port 0 (200OK), as suggested by rfc3264.<br><br></div>Regards,<br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <span dir="ltr"><<a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>Hello,<br></div><div>Is it possible to use FS as a recording server?</div><div><br></div><div>We will be receiving INVITEs with a siprec tag and sdp with the participants. To which FS should answer with 2 rtp ports on which to receive the audio. We would be receiving 2 rtp streams that we need to record to a file.</div><div><br></div><div>Thanks!</div><span><font color="#888888"><div><br></div><div>David</div></font></span></div>
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