<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">its not copying rtp packets, its copying decoded audio.<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Oct 11, 2017, at 4:16 PM, Tom Hartnett <<a href="mailto:hartnett.tom@gmail.com" class="">hartnett.tom@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Thanks for the clarification Michael. I was working off this old post<div class=""><a href="http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-June/002414.html" class="">http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-June/002414.html</a><br class=""></div><div class="">Which I guess is wrong or outdated.</div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Oct 11, 2017 at 3:46 PM, Michael Jerris <span dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" class="">mike@jerris.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word" class="">eavesdrop is going to transcode. its copying the raw already decoded audio from one channel to the other then re-encoding it.<span class=""><div class=""><br class=""><div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class="">On Oct 11, 2017, at 3:36 PM, Tom Hartnett <<a href="mailto:hartnett.tom@gmail.com" target="_blank" class="">hartnett.tom@gmail.com</a>> wrote:</div><br class="m_4441827826703441542Apple-interchange-newline"><div class=""><div dir="ltr" class="">Sorry to beat a dead horse, but so you mind if I ask why? The info I could discern (which may or may not be accurate) was that eavesdropping uses a simple RTP copy mechanism, while conferencing would involve converting all audio back and forth to PCM for each user (as well as jitter buffers and other overhead for each participant). Is there something I'm missing?</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris <span dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" class="">mike@jerris.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word" class="">mod_conference would be at least just as efficient as that, probably more. The feature you are looking for doesn’t exist today. <span class=""><div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class="">On Oct 11, 2017, at 2:42 PM, Tom Hartnett <<a href="mailto:hartnett.tom@gmail.com" target="_blank" class="">hartnett.tom@gmail.com</a>> wrote:</div><br class="m_4441827826703441542m_-3640493835163033906Apple-interchange-newline"><div class=""><div dir="ltr" class="">Thanks for your comment Michael,<div class="">Would it be reasonable to have a bunch of callers (listeners) spy or eavesdrop on a single call to a dummy endpoint (sender)?</div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris <span dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" class="">mike@jerris.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">we currently do not have a media distribution method to other endpoint modules other than mod_conference. You can use one of the stream recording methods to distribute to cdn as long as the delay isn’t an issue.<br class="">
<div class=""><div class="m_4441827826703441542m_-3640493835163033906h5"><br class="">
> On Oct 11, 2017, at 12:38 PM, Tom Hartnett <<a href="mailto:hartnett.tom@gmail.com" target="_blank" class="">hartnett.tom@gmail.com</a>> wrote:<br class="">
><br class="">
> Greetings all,<br class="">
> I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming.<br class="">
> I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either.<br class="">
> Can anyone offer any advice? Maybe FS is just not suited to this?<br class="">
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