<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="">Hi Julien and Daniel (and everyone),<div class=""><br class=""></div><div class="">Thank you very much Julien for speaking about oRTP.</div><div class="">Our company, Belledonne Communications (the company backing the Linphone project), invested a lot of its R&D effort in improving the audio and video quality. </div><div class="">For more information about oRTP’s innovative jitter buffer algorithm, you can read our article online : <a href="http://www.linphone.org/news/introduction-our-new-rtp-adaptive-jitter-buffer-algorithm" class="">http://www.linphone.org/news/introduction-our-new-rtp-adaptive-jitter-buffer-algorithm</a></div><div class="">To know more about oRTP : <a href="http://www.linphone.org/technical-corner/ortp" class="">http://www.linphone.org/technical-corner/ortp</a></div><div class=""><br class=""></div><div class="">Do not hesitate to contact me directly if you have questions about oRTP and Linphone’s jitter buffer.</div><div class=""><br class=""></div><div class="">Best regards,</div><div class=""><br class=""></div><div class="">
<div style="color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div style="color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div style="color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Elisa Nectoux<br class=""><b class="">Sales & Marketing</b><br class=""><br class="">+33 (0)9 52 63 65 05<br class=""><a href="mailto:elisa.nectoux@belledonne-communications.com" class="">elisa.nectoux@belledonne-communications.com</a><br class=""></div></div></div>
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<br class=""><div style=""><blockquote type="cite" class=""><div class="">Le 11 avr. 2019 à 12:13, Lorenzo Miniero <<a href="mailto:lminiero@gmail.com" class="">lminiero@gmail.com</a>> a écrit :</div><br class="Apple-interchange-newline"><div class=""><div dir="auto" class="">I think I remember one being available in Speex as well, but I may be wrong.<div dir="auto" class=""><br class=""></div><div dir="auto" class="">Lorenzo</div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 11 Apr 2019, 09:34 Julien Chavanton, <<a href="mailto:jchavanton@gmail.com" class="">jchavanton@gmail.com</a>> wrote:<br class=""></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class=""><div dir="ltr" class="">There is an interesting adaptive jitter buffer in oRTP that was improved a few years ago, <div class="">it is the coded domain, dealing with RTP packets, so it will have to drop packets if it needs to accelerate. </div><div class=""><br class=""></div><div class="">The one in WebRTC "NetEQ" will to fast accelerate in the decoded domain.</div><div class=""><br class=""></div><div class="">Both of them are kind of tightly coupled to the application / library </div><div class=""><br class=""></div><div class="">Cheers,</div><div class="">Julien</div></div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Apr 10, 2019 at 12:35 PM Daniel Pocock <<a href="mailto:daniel@pocock.pro" target="_blank" rel="noreferrer" class="">daniel@pocock.pro</a>> wrote:<br class=""></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><br class="">
Hi everybody,<br class="">
<br class="">
We've been using sipXtapi as part of reSIProcate's librecon (User agent<br class="">
library). Unfortunately, the jitter buffer in the free version of<br class="">
sipXtapi is not ideal when there are occasional periods of high latency.<br class="">
When latency comes back to normal, the audio is being replayed with a<br class="">
fast-forwarding sound effect.<br class="">
<br class="">
I'd like to provide an alternative option in the packages that I'm<br class="">
building. Many other products include jitter buffers that we could borrow.<br class="">
<br class="">
Could anybody make any suggestion about which implementation to use?<br class="">
<br class="">
It needs to be C or C++<br class="">
<br class="">
Regards,<br class="">
<br class="">
Daniel<br class="">
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