search for: maxaveragebitrate

...ll", with VoIP-out obviously we want to tune it. I am planning to use fmtp in SDP to control server/client Opus settings. Something like : - *maxplaybackrate*: a hint about the maximum output sampling rate that the receiver is capable of rendering in Hz. maxplaybackrate=8000 - *maxaveragebitrate*: specifies the maximum average receive bitrate of a session in bits per second (b/s). maxaveragebitrate=12000 I have a question about the documentation : "RTP Payload Format for Opus Speech and Audio Codec draft-spittka-payload-rtp-opus-03 maxaveragebitrate : Any positive integer is...

Hello, Is FEC supposed to work in NARROWBAND mode ?(with maxaveragebitrate=12000; maxplaybackrate=8000 ) ?I am having some confusing results, it appears that FEC is enabled in the encoder, but the decoder cannot find any packet with FEC. I am also wondering if this piece of code is correct (webrtc): /* The following is to parse the LBRR flags. */? if (opus_packet_parse(p...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...8000 a=rtpmap:118 L16/16000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:119 speex/32000 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:96 SILK/8000 a=fmtp:96 maxaveragebitrate=10000 a=fmtp:96 usedtx=0 a=fmtp:96 useinbandfec=1 a=rtpmap:108 SILK/12000 a=fmtp:108 maxaveragebitrate=12000 a=fmtp:108 usedtx=0 a=fmtp:108 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109...

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

...o make the best decision on a packet-per-packet basis as to whether FEC is useful considering the bit-rate, percentage of loss, and the exact audio content. Cheers, Jean-Marc On 05/02/15 11:46 AM, Dragos Oancea wrote: > Hello, > > Is FEC supposed to work in NARROWBAND mode (with > maxaveragebitrate=12000; maxplaybackrate=8000 ) ? > I am having some confusing results, it appears that FEC is enabled in > the encoder, but the decoder cannot find any packet with FEC. > > I am also wondering if this piece of code is correct (webrtc): > > /* The following is to parse the LBRR fl...

Hi all, I try to enable FEC in the encoder using the macro OPUS_SET_INBAND_FEC and I set the packet loss percentage to a constant value of 30%, using the macro OPUS_SET_PACKET_LOSS_PERC. Please find my encoder settings below: opus: encoder fmtp (maxplaybackrate=8000;maxaveragebitrate=24000;sprop-stereo=1;cbr=1;useinbandfec=1;usedtx=1) opus: encode bw=narrow bitrate=24000 fch=auto vbr=0 fec=1 expected loss=30 dtx=1 complex=10 At the decoder side when a packet is lost I call the decoder with the next params: opus_decode(ads->dec, NULL, 0, sampv, (int)(*sampc/ads->ch),...