search for: minptime

...-256 CF:30:A7:7F:71:11:D4:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43 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=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mU...

...tpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:2003565451 cname:0Cqf6EiGG5oFoWF5 a=ssrc:2003565451 msid:dXVhxyOSxULu3iClZayhTeEBzH2vob...

...0000000] chan_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....

...params:rtp-hdrext:ssrc-audio-level ????a=sendrecv ????a=mid:audio ????a=rtcp-mux ????a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:haR/UikskQr/AIrry5udqINI1hYfc5zY2I6jrkKT ????a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:waQfKIHI9UyjPVI0vrcUREDbSVZdtfCtRQK71/Ks ????a=rtpmap:111 opus/48000/2 ????a=fmtp:111 minptime=10 ????a=rtpmap:103 ISAC/16000 ????a=rtpmap:104 ISAC/32000 ????a=rtpmap:0 PCMU/8000 ????a=rtpmap:8 PCMA/8000 ????a=rtpmap:107 CN/48000 ????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:2121187131 cname:RXEE...

...:7C:8E:1F:2A: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:369615148...

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,

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

...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=...

...:24] a=setup: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...

...:audio [Aug 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...

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

...pmap: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:108 useinbandfec=1 a=rtpmap:109...

...params:rtp-hdrext: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...

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