search for: h264

Here's another problem I have with the 0.24 version of ffmpeg2theora. When I try to convert a h264 file to theora... (Note that for size and runtime reasons, foo.mts is a truncated file, I just took the first 32MB of the original file) ffmpeg2theora-0.24.linux32.bin foo.mts -x 1280 -y 720 -o foo-ffmpeg2theora-0.24.ogv Input #0, mpegts, from 'foo.mts': Duration: 00:00:15.83, start: 1.0...

Hi, I'm using the Record dialplan Application in an Context. My goal is to get a single screenshot of the h264 media stream per call. same => n,Record(/tmp/test.wav,0,10,qk) I nicely get a File test.h264. Is there a way to Playback this h264 video file on my computer or convert it somehow? VLC can't take it somehow. Regards, Benjamin -------------- next part -------------- An HTML attachment was...

...nd specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course), and I do specify also the Codecs: h264,h263 etc... Asterisk reports: *File /tmp/video does not exist in any format *>* Unable to open /tmp/video (format ulaw|h263|h264)* Looking then at the code and attaching with the debugger. the ast_openstream_full() function has this condition: if (!fileexists_core(filename, NULL, pr...

I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems with the format H264, Asterisk 12.8.1 compiled on the same hardware is behaving very well for the same format H264 Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality. Could someone investigate the problem of Asterisk 13 with video support on H264 ? Thank you. -------------- next part -------------- A...

...file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg0000 (format (ulaw|h264)): No such file or directory [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: app_voicemail.c:8609 play_message: Playback of message /var/spool/asterisk/voicemail/default/2036/INBOX/msg0000 failed [Apr 7 16:05:50] -- <SIP/2036-000000ee> Playing 'vm-advopts.gsm' (language 'en'...

Hello, We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good leads by google:ing, but of course I may have missed it Any pointers? BR, Torbj?rn Abrahamsson -------------- next part -------------- An HTML...

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 This is de sip debug on INVITE (Ekiga calls GXV3140) : v=0 o=grandstream 8000 8000 IN IP4 192.168.1.103 s=SIP Call c=IN IP4 192.168.1.103...

...from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video Support: Yes   Prim.Transp. : WS   Allowed.Trsp : WSS   SIP Options  : (none)   Codecs       : (opus|h264)   Status       : OK (75 ms)   Useragent    : SIP.js/0.12.0   Reg. Contact : sip:llghjqha at 192.0.2.239;transport=wss   RTP Engine   : asteri...

Hi all, I have to compare the theora codec with MPEG and H264. I was googling and I found that the PSNR is a common used parameter. How can I do this with Theora? Thanks -- Leonardo de Paula Rosa Piga Undergraduate Computer Engineering Student LSC - IC - UNICAMP http://www.students.ic.unicamp.br/~ra033956

...at lists.digium.com] Im Auftrag von Benjamin Marty Gesendet: Mittwoch, 11. April 2018 08:55 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video I think I found the root cause. The H264 Early Media video is received successfully on the Asterisk Server. It also seems to get processed. But it's send to the private IP of the receipent SIP phone. For clarification: 178.82.XX.XX is my Public IP of my Internet access. Both phones use this as Public IP via standard Source NAT. 159.89...

Link here http://news.zdnet.com/2100-9595_22-318208.html Will H264 become web standard in the future?

Link here http://news.zdnet.com/2100-9595_22-318208.html Will H264 become web standard in the future?

...Public IP. If I make a call from > an Early Media (video&audio) capable device to an Early Media capable > device (also video&audio) the Early Media audio works perfectly. But no > video. If I sniff with wireshark on the recipent device I just see G711 > (audio) RTP traffic. The h264 RTP traffic is missing before I accept the > call. After accepting the call the h264 RTP traffic comes through. > > The 183 SIP protocoll comes through. Even Asterisk is noticing it: > -- PJSIP/6002-00000013 is making progress passing it to PJSIP/6001-00000012 > > I tried both Ast...

Here is something that could change a lot of thinks on video/ audio streaming on the web http://www.kaourantin.net/2007/08/what-just-happened-to-video-on-web_20.html a shame they did not add theora/vorbis. -- %<------------------------------------------------------->% Michel memeteau VOIP | Visio: sip:freechelmi@gizmoproject.com Fixe : 0491886375 !!! APARTIR du 15/09/2007 ->

Here is something that could change a lot of thinks on video/ audio streaming on the web http://www.kaourantin.net/2007/08/what-just-happened-to-video-on-web_20.html a shame they did not add theora/vorbis. -- %<------------------------------------------------------->% Michel memeteau VOIP | Visio: sip:freechelmi@gizmoproject.com Fixe : 0491886375 !!! APARTIR du 15/09/2007 ->

...odd A CIV NAVSURFWARCENDIV Crane wrote: > If anyone has clues on how to build and add just a portion of > gst-plugins-bad into the existing EL OS set of gstreamer plugins, please > share (even URL pointers). > > [1] http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/openh264 > [2] https://en.wikipedia.org/wiki/OpenH264 > [3] http://www.openh264.org/ > > Even when this disclaimer is not here: > I am not a contracting officer. I do not have authority to make or modify the terms of any contract. All we did was rebuild the CentOS 6 packages tweaking the SPE...

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000...

I agree I have not seen any H264 streaming info for icecast2. If you are however interested in using ogg/vorbis (Theora) look up Theora Streaming Studio in the Theora Cookbook and it will give you some info. Additionally if you wish to script this task you can look up the info on the three commands TSS wraps, specifically dvgrab/...

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is...

...igium.com] On Behalf Of Fran Sent: Wednesday, 11 January 2006 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] video development I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SER&...