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Brief Introduction of SIP and SDP Protocol – Yeastar …
https://support.yeastar.com/hc/en-us/articles/360009806434-Brief-Introduction-of-SIP-and-SDP-Protocol-
SDP parameters. IP PBX and IP Phone use SIP to establish calls and use SDP to negotiate the parameters of media stream (audio, video). Here are some related parameters in SDP Media description. m=media name, port, proto and payload; media name: audio, video. port: the port to receive media stream. proto: RTP/AVP, RTP/SAVP. RTP/AVP represents RTP
The Session Description Protocol (SDP)
https://www.3cx.com/blog/voip-howto/sdp-voip2/
The Session Description Protocol (SDP)-VoIP Part 2. When a SIP based VoIP call is established, the audio or video sent between two SIP entities or more is streamed. Since many different codecs are supported by different devices or software, and each individual SIP entity taking part in the call does not know the IP address of the other SIP ...
SIP - Session Description Protocol - Tutorialspoint
https://www.tutorialspoint.com/session_initiation_protocol/session_initiation_protocol_sdp.htm
SDP is generally contained in the body part of Session Initiation Protocol popularly called SIP. SDP is defined in RFC 2327. An SDP message is composed of a series of lines, called fields, whose names are abbreviated by a single lower-case …
Understanding Media In SIP Session Description Protocol ...
https://teraquant.com/understand-media-sip-session-description-protocol/
There are many SIP networks where calls either fail or there are issues with a conference bridge where clients offering a diversity of codecs cause failed negotiations or one-way audio. In addition, the increased use of VPN especially from current traditional SD-WAN implementations cause incorrect mapping of IP addresses and increase latencies. Video and …
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
Do you have the SIP traces showing no audio codec? I am surprised a carrier would not include any m=audio or a=rtpmap if they are sending the invite with SDP. I believe the default codec for a voip dial peer is g.729, but as long as you have a voice class codec assigned to the dial peer with the other codecs, that should be fine.
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