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sip - SDP Offer/Answer model with DTMF rtpmap/fmtp ...

    https://stackoverflow.com/questions/49848381/sdp-offer-answer-model-with-dtmf-rtpmap-fmtp-mismatch
    m = audio 1235 RTP/AVP 8 120 a = rtpmap:8 PCMA/8000 a = rtpmap:120 telephone-event/8000 a = fmtp:101 0-15 a = sendrecv From RFC 3264: For streams marked as sendrecv in the answer, the "m=" line MUST contain at least one codec the answerer is willing to both send and receive, from amongst those listed in the offer.

UNDERSTANDING SIP TRACES - Cisco Community

    https://community.cisco.com/t5/collaboration-voice-and-video/understanding-sip-traces/ta-p/3137704
    m=audio 11164 RTP/AVP 18 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no +++From our understanding of the traces, we see that the call originates from a device called Broadworks, which advertises G711a, G711u, G729 and uses rtp-nte for DTMF.

Real-Time Transport Protocol (RTP) Parameters

    https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml
    In addition, for the payload formats listed in the RTP Payload Types table above, the "encoding name" is also registered as a media subtype under the media type "audio" or "video". The clock rate and number of channels shown here are the normal values for those payload formats that have a normal value.

Understanding Media In SIP Session Description Protocol ...

    https://teraquant.com/understand-media-sip-session-description-protocol/
    m=audio 49230 RTP/AVP 18 96 97 98 a= rtpmap:18 G729/8000/1 a=rtpmap:96 L8/8000 a=rtpmap:97 L16/8000 a=rtpmap:98 L16/11025/2 a=fmtp:18 annexb=yes. During the early stages of RTP development, statically assigned payload types (0-34) were used to bind encodings to payload types.

The Session Description Protocol (SDP)

    https://www.3cx.com/blog/voip-howto/sdp-voip2/
    m=audio 61896 RTP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv. From the above capture’s we can see that: Session Description section. Version - V=0. SDP Protocol Version. Up to now there are no minor versions.

RTP Payload Format and File Storage Format for Opus Speech ...

    https://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.html
    m=audio 54312 RTP/AVP 100 a=rtpmap:100 opus/48000 Opus supports several clock rates. For signaling purposes only the highest, i.e. 48000, is used. The actual clock rate of the corresponding media is signaled inside the payload and is not subject to this payload format description.

RFC 6871 - Session Description Protocol (SDP) Media ...

    https://tools.ietf.org/html/rfc6871
    The "m=" line and the "rtpmap" attribute offer an audio stream and provide the lowest precedence configuration (PCMU without any DTMF encoding). The rmcap lines define the RTP- based media format capabilities (PCMU, G729, telephone-event, H263-1998, and H264) and the omcap line defines the non-RTP-based media format capability (wildcard). ...

Understanding DTMF negotiation and troubleshooting on SIP ...

    https://community.cisco.com/t5/collaboration-voice-and-video/understanding-dtmf-negotiation-and-troubleshooting-on-sip-trunks/ta-p/3152511
    a=label:main-audio. a=rtpmap:114 x-msrta/16000. a=fmtp:114 bitrate=29000. a=rtpmap:115 x-msrta/8000. a=fmtp:115 bitrate=11800. a=rtpmap:112 G7221/16000. a=fmtp:112 bitrate=24000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:116 AAL2-G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. …

RFC 4566 - SDP: Session Description Protocol

    https://tools.ietf.org/html/rfc4566
    RFC 4566 SDP July 2006 1.Introduction When initiating multimedia teleconferences, voice-over-IP calls, streaming video, or other sessions, there is a requirement to convey media details, transport addresses, and other session description metadata to the participants. SDP provides a standard representation for such information, irrespective of how that information is transported.

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