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The Session Description Protocol (SDP) - 3CX
https://www.3cx.com/blog/voip-howto/sdp-voip2/#:~:text=SIP%20phone%20A%20has%20the%20following%20codec%20priority%3B,first%20matched%20codec%20between%20the%202%20SIP%20phones.
SIP Protocol: What Is & How It Works in a VOIP Call ...
https://www.softwareadvice.com/resources/what-is-sip/
G.711 codec: Used for uncompressed digital voice. Audio quality is better than other codecs, but it uses more bandwidth. G.729 codec: Used for compressed voice. It lowers the audio quality to reduce the amount of transmitted data and the resulting bandwidth consumption. Encoded packets of audio data are carried by the real-time transport protocol (RTP), a …
Appendix C: Video and Audio Codecs used by H.323 and …
https://www.c21video.com/technical-papers/skype-for-business/appendix-c--h-323-and-sip-video-codecs
Also known as Pulse Code Modulation (PCM), G.711 is a commonly used audio codec were the 300-3400 Hz analogue audio is encode at a rate of 8000Hz to provide toll-quality audio in a 64 kbps stream. There are two versions, PCMU (µ-law) is mainly used in North America and PCMA (A-law) which is used in most other countries.
SIP Trunking For Dummies: Which Codec Should You Use ...
https://teledynamic.com/2013/05/sip-trunking-for-dummies-which-codec-should-you-use/
Common VoIP Codec Protocols G.729 G.729 is a codec that has low bandwidth requirements but provides good audio quality. This is the most commonly used codec in VoIP calling and has a MOS rating of 4.0 G.711 G.711 is a codec that was introduced by ITU in 1972 for use in digital telephony.
The World's Audio Network - sip.audio
Audio Codecs: OPUS (Direct, Relayed, ISDN*) G711 (Direct, Relayed, POTS*)) G722 (Direct, Relayed, ISDN*) MPEG-1 (Direct) MPEG-2 LII (Direct, ISDN 128 Mono+JS, 64*) MPEG-4 AAC (Direct, ISDN 128*) MPEG-4 AAC LD (Direct, ISDN 64 & 128*) MPEG-4 HE-AACv2 (Direct) MPEG-1/2 Layer III (Direct, ISDN 128*) PCM (Direct) Enhanced APT-X (Direct) iLBC (Direct, Relayed)
codec values in SDP - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/codec-values-in-sdp/td-p/3035212
8811 SIP phone--CUCM--isr4331 pstn gw. Media peers: 8811 SIP phone -- isr4331 pstn gw. In debug voip ccapi inout I see on ISR 4331 that call codec for media is 0x4 (g729) instead of 0x8 (g729A). I set voice card codec complex to medium but remained the same. 8811 advertises (from device pool) g729A, g729 and g729B.
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.
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