search for: useinbandfec

...> On Tue, Mar 3, 2015, 12:58 AM Kelvin Chua <kelchy at gmail.com> wrote: > >> Hi guys, >> >> I have been reading a lot about the "adaptiveness" of opus and i quote: >> >> ... can still change, e.g. to adapt to changing network conditions. >> useinbandfec ... >> >> can somebody please enlighten me on this "adaptiveness"? >> whatever way I do our tests, it sticks to the same sampling rate and the >> same average bitrate, it would go up, down a bit but that's it. >> When we get some network issues, bandwidth...

...er-Agent: CSipSimple_d2vzw-16/r1916 Content-Type: application/sdp Content-Length: 354 v=0 o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1 t=0 0 m=audio 4006 RTP/AVP 96 3 0 8 101 c=IN IP4 172.17.9.1 a=rtcp:4007 IN IP4 172.17.9.1 a=sendrecv a=rtpmap:96 SILK/8000 a=fmtp:96 useinbandfec=0 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 16 lines) --- Sending to 172.17.9.1 : 55388 (NAT) Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS <--- Reliably Tran...

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

...over a wide range. > > On Tue, Mar 3, 2015, 12:58 AM Kelvin Chua <kelchy at gmail.com> wrote: > > Hi guys, > > I have been reading a lot about the "adaptiveness" of opus and i quote: > > ... can still change, e.g. to adapt to changing network conditions. > useinbandfec ... > > can somebody please enlighten me on this "adaptiveness"? > whatever way I do our tests, it sticks to the same sampling rate and the > same average bitrate, it would go up, down a bit but that's it. > When we get some network issues, bandwidth utilization stays t...

...=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 maxaveragebitrate=20000 a=fmtp:109 usedtx=0 a=fm...

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

Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"? whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it. When we get some network issues, bandwidth utilization stays the same. Am I interpreti...

...es support continuously varying the bitrate over a wide range. On Tue, Mar 3, 2015, 12:58 AM?Kelvin Chua <kelchy at gmail.com> wrote: Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"?whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it.When we get some network issues, bandwidth utilization stays the same.Am I interpreting...

...n_sip.c: Processing media-level (audio) SDP a=rtpmap:111 opus/48000/2... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp-fb:111 transport-cc... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 minptime=10;useinbandfec=1... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:63 red/48000/2... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:63 111/111... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing me...

...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), 0); and set the flag packet_lost=true;...

...es support continuously varying the bitrate over a wide range. On Tue, Mar 3, 2015, 12:58 AM?Kelvin Chua <kelchy at gmail.com> wrote: Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"?whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it.When we get some network issues, bandwidth utilization stays the same.Am I interpreting...

...F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw a=ssrc:3696151487 msid:cC3clldcCI...

...:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24...

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

...1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=sendrecv > a=rtcp-mux > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10; useinbandfec=1 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:3696151487 c...

...9 22:15:50] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [Aug 9 22:15:50] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [Aug 9 22:15:50] a=sendrecv [Aug 9 22:15:50] a=rtcp-mux [Aug 9 22:15:50] a=rtpmap:111 opus/48000/2 [Aug 9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1 [Aug 9 22:15:50] a=rtpmap:103 ISAC/16000 [Aug 9 22:15:50] a=rtpmap:104 ISAC/32000 [Aug 9 22:15:50] a=rtpmap:9 G722/8000 [Aug 9 22:15:50] a=rtpmap:0 PCMU/8000 [Aug 9 22:15:50] a=rtpmap:8 PCMA/8000 [Aug 9 22:15:50] a=rtpmap:106 CN/32000 [Aug 9 22:15:50] a=rtpmap:105 CN/16000 [Aug 9 22:15:50...

...drext:ssrc-audio-level [Aug 11 15:53:47] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [Aug 11 15:53:47] a=sendrecv [Aug 11 15:53:47] a=rtcp-mux [Aug 11 15:53:47] a=rtpmap:111 opus/48000/2 [Aug 11 15:53:47] a=rtcp-fb:111 transport-cc [Aug 11 15:53:47] a=fmtp:111 minptime=10;useinbandfec=1 [Aug 11 15:53:47] a=rtpmap:103 ISAC/16000 [Aug 11 15:53:47] a=rtpmap:104 ISAC/32000 [Aug 11 15:53:47] a=rtpmap:9 G722/8000 [Aug 11 15:53:47] a=rtpmap:0 PCMU/8000 [Aug 11 15:53:47] a=rtpmap:8 PCMA/8000 [Aug 11 15:53:47] a=rtpmap:106 CN/32000 [Aug 11 15:53:47] a=rtpmap:105 CN/16000 [Aug 11 15:53:47...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or