<instructions>Your goal is to analyze the following comment and estimate how highly it will be upvoted by the Hacker News community.</instructions><comment><author>tracerbulletx</author><text>In 2015 I wrote a whole Contact Management app for a business that let their entire sales team of like 50 reps pick up phone calls from the browser or from their desk phone using Twilios SIP trunk service. It was pretty cool, it routed the calls to both our internal Asterisk server, and worked flawlessly with Twilios WebRTC libraries in the web app.  It was really cool and surprisingly easy, also all of the call recordings instantly became available in the web app and they could initiate calls from their call calendar in the app.</text><parent_chain></parent_chain></comment><story><title>Show HN: Bring phone calls into the browser (SIP-to-WebRTC)</title><url>https://github.com/pion/example-webrtc-applications/tree/master/sip-to-webrtc</url></story>
<instructions>Your goal is to analyze the following comment and estimate how highly it will be upvoted by the Hacker News community.</instructions><comment><author>Sean-Der</author><text>Really exciting times for WebRTC in general right now!&lt;p&gt;If you are new to WebRTC and want to learn more about the protocol check out [0] (would love peoples feedback). It is used in lots of unexpected places like streaming (added to OBS)[1] and Embedded [2].&lt;p&gt;I am especially excited with new implementations popping up like [3] and [4].&lt;p&gt;[0] &lt;a href=&quot;http:&amp;#x2F;&amp;#x2F;webrtcforthecurious.com&quot; rel=&quot;nofollow&quot;&gt;http:&amp;#x2F;&amp;#x2F;webrtcforthecurious.com&lt;/a&gt;&lt;p&gt;[1] &lt;a href=&quot;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;Glimesh&amp;#x2F;broadcast-box&quot;&gt;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;Glimesh&amp;#x2F;broadcast-box&lt;/a&gt;&lt;p&gt;[2] &lt;a href=&quot;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;sepfy&amp;#x2F;libpeer&quot;&gt;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;sepfy&amp;#x2F;libpeer&lt;/a&gt;&lt;p&gt;[3] &lt;a href=&quot;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;algesten&amp;#x2F;str0m&quot;&gt;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;algesten&amp;#x2F;str0m&lt;/a&gt;&lt;p&gt;[4] &lt;a href=&quot;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;elixir-webrtc&amp;#x2F;ex_webrtc&quot;&gt;https:&amp;#x2F;&amp;#x2F;github.com&amp;#x2F;elixir-webrtc&amp;#x2F;ex_webrtc&lt;/a&gt;</text><parent_chain></parent_chain></comment><story><title>Show HN: Bring phone calls into the browser (SIP-to-WebRTC)</title><url>https://github.com/pion/example-webrtc-applications/tree/master/sip-to-webrtc</url></story>