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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
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 PCMU 8000 Hz, 8 is PCMA 8000 Hz, and 101 is payload type for DTMF digits sending. There are some links that can be ...
Solved: Inbound Calls not working - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/inbound-calls-not-working/td-p/4293629
c=IN IP4 100.201.0.74 t=0 0 m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3 c=IN IP4 100.201.0.74 m=image 0 udptl t38 c=IN IP4 100.201.0.74 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:320 a=T38FaxUdpEC:t38UDPRedundancy. 003261: *Feb 18 …
Knowledgebase Article: SmartMedia: SDP profile ...
https://www.patton.com/support/kb_art.asp?art=408
m=audio 0 RTP/AVP 0 8 4 18 98 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6300;annexa=yes a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 a=rtpmap:13 CN/8000. Another example to disable silence suppression explicitly in the SDP.
Toolpack:Profile SDP Description - TB Wiki
https://docs.telcobridges.com/mediawiki/index.php/Toolpack:Profile_SDP_Description
m=audio 0 RTP/AVP 18 4 8 0 96 97 98 99 13 a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:98 telephone-event/8000 a=rtpmap:99 GSM-FR/8000 a=fmtp:4 bitrate=6.3 a=fmtp:4 annexa=no This example is G.711 alaw with a 10ms packet (instead of the default 20ms) : m=audio 0 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:10
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.
Asterisk 18.4 Dtmf Problem - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/asterisk-18-4-dtmf-problem/90456
o=- 0 1 IN IP4 10.101.0.165 s=-c=IN IP4 10.101.0.165 t=0 0 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
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/
Refer to the exhibit. Which outgoing m-line SDP is sent to Cisco UCM after matching the appropriate dial peers via Cisco Unified Border Element? A. m=audio 16550 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1. B. m=audio 16550 RTP/AVP 18 0 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1.
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