search for: g7221

...ycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP...

...information: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp:119 0-15 m=video 3232 RTP/AVP 109 110 a=rtcp-fb:* ccm fir tmmbr a=rtpmap:...

...like nothing extra gets compiled when the with-ssl is used... With --with-ssl: [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject --disable-speex-codec --disable-speex-aec --disable-bcg729 --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec --disable-g722-codec --disable-g7221-codec --disable-opencore-amr --disable-silk --disable-opus --disable-video --disable-v4l2 --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc --without-external-pa --without-external-srtp --disable-resample --disab...

...Media Gateway 5.1?Allow-Events: talk?Accept: application/sdp?Privacy: none?X-IP-Info: 10.11.11.3??v=0?o=FreeSWITCH 1453083377 1453083378 IN IP4 Provider_IP_Address?s=FreeSWITCH?c=IN IP4 Provider_IP_Address?t=0 0?m=audio 28388 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13?a=rtpmap:98 AMR/8000?a=rtpmap:99 G7221/16000?a=fmtp:99 bitrate=32000?a=rtpmap:100 G726-32/8000?a=rtpmap:102 iLBC/8000?a=fmtp:102 mode=30?a=rtpmap:101 telephone-event/8000?a=fmtp:101 0-16?a=ptime:20?m=audio 29684 RTP/AVP 4 101 13?a=rtpmap:101 telephone-event/8000?a=fmtp:101 0-16?a=ptime:30?m=audio 21364 RTP/AVP 8 0 98 9 99 100 18 3 102 1...

...may not be optimal, but I really need it, even if I have to patch Asterisk Thanks for your help SDP send by Tandberg : -------------------------- v=0 o=tandberg 1 5 IN IP4 192.168.50.10 s=- c=IN IP4 192.168.50.10 b=CT:1920 t=0 0 m=audio 48260 RTP/AVP 100 101 9 8 0 102 b=TIAS:64000 a=rtpmap:100 G7221/16000 a=fmtp:100 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-15 a=sendrecv m=video 48262 RTP/AVP 97 98 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:1920000 a=rtpmap:97 H264-RC...

...compiled when the with-ssl is used... > With --with-ssl: > [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject > --disable-speex-codec --disable-speex-aec --disable-bcg729 > --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec > --disable-g722-codec --disable-g7221-codec --disable-opencore-amr > --disable-silk --disable-opus --disable-video --disable-v4l2 > --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv > --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc > --without-external-pa --without-external-srtp --disa...

...726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=rtpmap:9 G722/8000 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=rtpmap:107 opus/48000/2 a=ptime:20 a=maxptime:20 a=sendrecv m=video 11188 RTP/AVP 31 34 103 99 104 100 108 a=ice-ufrag:28e29ae16a4cc8d54ebf03cf56010f1a a=ice-pwd:57ec01cf24ac689d2fb459bd0411d2...

...th-ssl is used... >> With --with-ssl: >> [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject >> --disable-speex-codec --disable-speex-aec --disable-bcg729 >> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec >> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr >> --disable-silk --disable-opus --disable-video --disable-v4l2 >> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv >> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc >> --without-external-pa --without-exte...

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ?? Why would you want to do this? Several reasons: - Predictability: When built with the ?bundled pjproject, you're always certain of the version you're running against, no matter where it's

...NVITE from windows RTC: v=0. o=- 0 0 IN IP4 172.31.9.130. s=session. c=IN IP4 172.31.9.130. b=CT:1000. t=0 0. m=audio 4632 RTP/AVP 97 111 112 6 0 8 4 5 3 101. k=base64:ftJemQZ2pTDV5gzzqxG6ps5Ol5qiOt2qbP9L9Or5JQg. a=rtpmap:97 red/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:112 G7221/16000. a=fmtp:112 bitrate=24000. a=rtpmap:6 DVI4/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=encryption:optional. a=direction:active. OK from asterisk 1.6 PBX: v=0. o=PBX...

...at are in .wav format. But when it tries to play any files with .siren14 extensions, I get complete noise coming out. Here's the negotiated SDP: v=0 o=root 1668560220 1668560220 IN IP4 207.10.184.50 s=Asterisk PBX 10.7.1 c=IN IP4 207.10.184.50 t=0 0 m=audio 16204 RTP/AVP 115 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=sendrecv If I rename away the .siren14 files, all is OK. I can't find anything related to this with a search.

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The

.../8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:108 G7221/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 29 lines) --- Sending to 192.168.96.141:5060 (no NAT) Sending to 192.168.96.141:5060 (no NAT) Using INVITE request a...

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

...es > [pjproject] Configuring with --prefix=/opt/pjproject > --with-external-speex --with-external-gsm --with-external-srtp > --with-external-pa --disable-video --disable-v4l2 --disable-sound > --disable-resample --disable-opencore-amr --disable-ilbc-codec > --without-libyuv --disable-g7221-codec --enable-epoll > aconfigure: error: Unable to use PortAudio. If PortAudio development > files are not available in the default locations, use CFLAGS and LDFLAGS > env var to set the include/lib paths > Makefile:57: recipe for target 'build.mak' failed > make: *** [build...

...p:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap: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=...

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

...lse would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps G7221_C.24kbps G7221.24kbps G7221_C.32kbps G7221.32kbps G7221_C.48kbps I can get Asterisk to work with G722 and the sound is superior compared to uLAW. I tried to get it working with Siren7 and Siren14 but I cannot. It always says i...

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