We have collected the most relevant information on Content-Type Audio/Telephone-Event. Open the URLs, which are collected below, and you will find all the info you are interested in.
Signaled Telephony Events in the Session Initiation ...
https://www.softarmor.com/wgdb/docs/draft-mahy-sipping-signaled-digits-01.html
This package re-uses the audio/telephone-event MIME type defined in RFC2833. Each telephone-event consists of a four octet structure. Multiple telephone-events MAY be concatenated. The structure of the audio/telephone-event MIME type is reproduced here from RFC2833 for the convenience of the reader.
DTMF Events through SIP Signaling - Cisco
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_proto/configuration/15-mt/cube-proto-15-mt-book/voi-dtmf-sip.pdf
Content-Type: audio/telephone-event Content-Length: 4 Configuring DTMF Events through SIP Signaling ToconfiguretheDTMFEventsthroughSIPSignalingfeature,performthefollowingsteps. SUMMARY STEPS 1. enable 2. configureterminal 3. sip-ua 4. timersnotifynumber 5. retrynotifynumber 6. exit DETAILED STEPS Command or Action Purpose
CM: SIP SDP Telephone-event codec clock rate mismatch - Avaya
https://support.avaya.com/public/index?page=content&id=SOLN334984&actp=LIST
"telephone-event", the media type as "audio/telephone-event". In accordance with current practice, this payload format does not have a static payload type number, but uses an RTP payload type number established dynamically and out-of-band. The default clock frequency is 8000 Hz, but the clock frequency can be redefined when assigning
RFC 4733 - RTP Payload for DTMF Digits, Telephony Tones ...
https://tools.ietf.org/html/rfc4733
RFC 4733 Telephony Events and Tones December 2006 For example, if the payload format uses the payload type number 100, and the implementation can handle the DTMF tones (events 0 through 15) and the dial and ringing tones (assuming as an example that these were defined as events with codes 66 and 70, respectively), it would include the following description in its SDP …
Understanding SIP DTMF Options supported by CUCM - Cisco
https://community.cisco.com/t5/collaboration-voice-and-video/understanding-sip-dtmf-options-supported-by-cucm/ba-p/3107307
Content-Type: audio/telephone-event. Content-Length: 4. 12/15/2011 09:50:01.879 CCM|getUnsolNotifyContents: Unsol NOTIFY message body is 0x040001f4, len=4. 12/15/2011 09:50:01.879 CCM|getUnsolNotifyContents: getUnsolNotifyContents: Parsed digit=4, duration=500, end flag=0, duplicateEvent=0
DTMF Issue in SIP - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/dtmf-issue-in-sip/td-p/2598977
Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 798-a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 . 2. +++CUCM determines the DTMF caps of both (unity connection and sip trunk, so it can send it in its SDP answer+++
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones ...
https://tools.ietf.org/html/rfc2833
RFC 2833 Tones May 2000 The RTP payload format is designated as "telephone-event", the MIME type as "audio/telephone-event". The default timestamp rate is 8000 Hz, but other rates may be defined. In accordance with current practice, this payload format does not have a static payload type number, but uses a RTP payload type number established dynamically and out-of-band.
Add a content type to a list or library
https://support.microsoft.com/en-us/office/add-a-content-type-to-a-list-or-library-917366ae-f7a2-47ad-87a5-9689a1884e60
In the Add Content Type page, select the Choose content type menu, and then select the content type you want to add, from the list of custom content types that can be added. Details about the selected content type will display, such as its description and column information. When you finish selecting the content type that you want to add ...
The Anatomy of an INVITE Request - Tao, Zen, and …
https://andrewjprokop.wordpress.com/2014/04/21/the-anatomy-of-an-invite-request/
Content-Type. Content-Type: application/sdp. If there is a message body, Content-Type will tell you how that message body has been formatted. In this case, it contains SDP. User-Agent. User-Agent: Avaya SIP Softphone. User-Agent is used to inform the calling party about the caller’s SIP endpoint. A good analogy is an Internet browser.
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