{"id":319,"date":"2016-11-13T10:01:56","date_gmt":"2016-11-13T09:01:56","guid":{"rendered":"http:\/\/www.dmedia.ch\/?post_type=portfolio&p=319"},"modified":"2025-02-12T11:26:57","modified_gmt":"2025-02-12T10:26:57","slug":"studiophone","status":"publish","type":"portfolio","link":"https:\/\/www.micromedia.ch\/?portfolio=studiophone","title":{"rendered":"Studiophone"},"content":{"rendered":"

Studiophone<\/h1>\n<\/div><\/section><\/div>\n
<\/div><\/div><\/div><\/div>\n

Professional SIP-IP-phone software for radio and TV production.<\/p>\n

The Studiophone is a modern software telephone designed specifically for broadcast applications. Operation is intuitive via mouse or touch screen. The Studiophone is a pure software client according to the international SIP standard, which can be used with a variety of compatible telephone exchanges (PBX), routers and gateways.<\/p>\n<\/div><\/section><\/div>\n

\n
\n
\n

Features<\/h3>\n

– Integrated conference with a maximum of 30 simultaneous participants in the standard version.
\n– Call Recording in 48 kHz
\n– Phone book
\n– SIP 2.0 Standard<\/p>\n

Codec:
\n– G711
\n– G722
\n– OPUS Stereo
\n– Linear 32kHz
\n– Linear 48kHz<\/p>\n

Operating systems:
\n– Windows 10, Windows 11
\n– macOS on request<\/p>\n

Soundcards:
\nWindows Standard WASAPI, USB-Interface or AoIP (Audinate Dante DVS or Ravenna)<\/p>\n<\/div>\n<\/div>\n<\/div>\n<\/div><\/section><\/div>

\n
\n
\n

Connectivity<\/h3>\n

All major SIP-PBX or SIP-Provider<\/p>\n

– Ascotel\/Aastra Intelligate
\n– Siemens HiPath
\n– Cisco CallManager
\n– Asterisk PBX
\n– Yeastar<\/p>\n

Compatibility:
\n– SIP 2.0
\n– N\/ACIP<\/p>\n<\/div>\n<\/div>\n<\/div>\n<\/div><\/section><\/div><\/p>\n

\n
\n
\n

Use cases<\/h3>\n

Studio telephony
\nListeners call the studio and the call will be broadcasted live or recorded.<\/p>\n

Newsroom
\nJournalists can record interviews in the newsroom or in the studio and load recordings directly into the audio editor. Compatible with most automation systems.<\/p>\n

Outside Broadcast
\nHigh Quality via OPUS enabled SIP server.<\/p>\n<\/div>\n<\/div>\n<\/div>\n<\/div><\/section><\/div>\n

<\/div><\/div><\/div><\/div>\n
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