search for: sprop

Displaying 6 results from an estimated 6 matches for "sprop".

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2015 Mar 23
2
PJSIP - Video Support for WebRTC
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
2019 Jul 15
0
How to enable OPUS inband FEC
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), 0); and set...
2010 Dec 06
1
Asterisk 1.6.2.10 video call
...pmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 35878 RTP/AVP 34 100 99 b=AS:128 a=sendrecv a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2; QCIF=2 a=rtpmap:100 H263-1998/90000 a=fmtp:100 CIF=2; QCIF=2 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== [Dec 6 15:11:18] Using INVITE request as basis request - 1666548288-45310-6 at BJC.BGI.B.BAD [Dec 6 15:11:18] Found peer 'grandstream' for 'grandstream' from 192.168.1.103:45310 [Dec 6 15:11:18] Found RTP audio format 8 [Dec 6 15:11:18] Fo...
2014 Dec 11
2
PJSIP configuration question
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
2014 Dec 11
0
PJSIP configuration question
...:112 AAL2-G726-32/8000 a=rtpmap:9 G722/8000 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:10...
2017 Jun 13
7
[Bug 2729] New: Can connect with MAC hmac-sha1 even though it's not configured on the server
https://bugzilla.mindrot.org/show_bug.cgi?id=2729 Bug ID: 2729 Summary: Can connect with MAC hmac-sha1 even though it's not configured on the server Product: Portable OpenSSH Version: 7.5p1 Hardware: All OS: Linux Status: NEW Severity: security Priority: P5