<div dir="ltr">Hi!<div><br></div><div>From my experience, verto is much more stable and more resource and browser(js) friendly, than sipjs and other SIP implementations of SIP.</div><div>With various SIP libraries I’ve got strange issues, like sound delay (time to time), not working audio in most fresh browsers, other some strange issues. But verto was installed over a year ago and still working without any troubles (with FS 1.5, I believe)</div><div>So, my voice is for verto.</div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-04-20 11:08 GMT+03:00 Oivvio Polite <span dir="ltr"><<a href="mailto:mylists@polite.se" target="_blank">mylists@polite.se</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Summary:<br>
<br>
I'm just getting started with VoIP/SIP/WebRTC. Following the FreeSwitch<br>
Cookbook I've built a webclient that uses SIP signalling to establish a<br>
connection with FreeSwitch and then exchanges media with WebRTC. But I<br>
get the impression from a lot of sources that SIP is a bad fit for<br>
WebRTC. One datapoint that indicates this is the very existance of<br>
mod_verto. Why would you bother writing it if SIP, that has 10+ years of<br>
battle testing, was a good fit right?<br>
<br>
What I don't understand yet is why SIP is a bad fit for WebRTC? Would<br>
someone in the know care to muse about that?<br>
<br>
regards Oivvio<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr">Best regards,<div>Igor</div></div></div>
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