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      <title>Self-Hosted VoIP Media Processing: FreeSWITCH vs RTPEngine vs Asterisk Codec Transcoding Guide</title>
      <link>https://www.pistack.xyz/posts/2026-06-03-self-hosted-voip-media-codec-transcoding-freeswitch-rtpengine-guide/</link>
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      <description>&lt;h2 id=&#34;introduction&#34;&gt;Introduction&lt;/h2&gt;&#xA;&lt;p&gt;VoIP media processing — the real-time conversion of audio codecs between different endpoints — is one of the most CPU-intensive workloads in telecommunications infrastructure. When a SIP call connects a G.711 leg to an Opus leg, or bridges WebRTC (Opus) to a traditional PSTN trunk (G.711), a media processor must transcode the audio in real time with sub-20ms latency to maintain call quality.&lt;/p&gt;</description>
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