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UNDERSTANDING SIP TRACES - Cisco Community
https://community.cisco.com/t5/collaboration-voice-and-video/understanding-sip-traces/ta-p/3137704
m=audio 25268 RTP/AVP 18 0 8 101 . This line defines the media attribtes that will be used for the call. Audio: means that this is an Audio call, we …
Description of Media parameter in SIP "m=audio 12548 …
https://stackoverflow.com/questions/27869523/description-of-media-parameter-in-sip-m-audio-12548-rtp-avp-0-8-101
This answer is not useful. Show activity on this post. It's a field from SDP protocol, describing parameters of media ("m" is for "media"). Next, type of media is "audio", not video, for example. (m=audio). 12548 is a port address for streaming media. "RTP/AVP" means "RTP Audio/Video Profile" and representing one of RTP profiles, which are coded by 0, 8 and 101. 0 is …
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.
Modifying the m and a lines in SDP using SIP Message ...
https://support.sonus.net/display/UXDOC70/Modifying+the+m+and+a+lines+in+SDP+using+SIP+Message+Manipulation
m=audio 28048 RTP/AVP 18 116 c=IN IP4 41.87.234.68 a=rtpmap:18 G729/8000 a=rtpmap:116 telephone-event/8000 As mentioned in the About This Page , the device that originated the above SDP will still expect to receive telephone-events as RTP event 116.
SIP understanding debug and traces - DrVoIP
https://drvoip.com/support/knowledgebase.php?article=52
m=audio 25268 RTP/AVP 18 0 8 101 . This line defines the media attribtes that will be used for the call. Audio: means that this is an Audio call, we can also have m=video in case of a Video call. 25268: Is the dynamic RTP port used for the call. RTP/AVP: Represents the RTP/AVP profile number for each of the profiles listed.
m=audio 20040 RTP/AVP 18 0 8 101 102 2 103 4 104 3 …
https://pastebin.com/5iQUUETf
m=audio 20040 RTP/AVP 18 0 8 101 102 2 103 4 104 3 105 106 107 125 99 100
Solved: SIP/SDP - no codec specifed - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/sip-sdp-no-codec-specifed/td-p/3913990
The numbers at the end of this m line: m=audio 23610 RTP/AVP 8 0 18 97 Those are the codecs. They are RTP payload types, and in preference order. Eg 8=g711alaw, 0=g711ulaw, 18=g729, 97=this one is special, it's actually what they want to Mark RTP-NTP (RFC2833) DTMF relay as.
Asterisk 18.4 Dtmf Problem - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/asterisk-18-4-dtmf-problem/90456
m=audio 20546 RTP/AVP 18 8 96 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=sendrecv a=ptime:20 <— Transmitting SIP response (1169 bytes) to UDP:10.101.0.122:5060 —> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.101.0.122:5060;rport=5060;received=10.101.0.122;branch=z9hG4bK116.9ed545c4a9077aa619bc8defadfc3c8d.0
Exam 350-801 topic 1 question 73 discussion - ExamTopics
https://www.examtopics.com/discussions/cisco/view/55494-exam-350-801-topic-1-question-73-discussion/
From ITSP SDP- m-audio 5000 RTP/AVP 18 0 -> first preference G729, 2nd G711ulaw Calls from ITSP -> match dial-peer 13 -> voice class codec 1 - > first preference G711alaw it will match m=audio 16500 RTP/AVP 8 0 18 Correct Answer: C
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