search for: h263

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...

By browsing on the mailing list I learned that its possible to generate .h263 asterisk friendly files with gstreamer. The script below it's supposed to do just that, however I get error when trying it out locally. gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux ! ffdec_h263 ! videoscale ! video/x-raw-yuv,width=352,height=288 ! ffenc_h263 rtp...

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show channel ...' command shows there is only audio channel, no video channel. I don't understand why. The file I tried to play is $ ls -al /var/lib...

I'm trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP). Problem is my video system only speaks H263-2000 version (aka H263++). As far as I can see, * only understands H263 and H263+ in and sdp. Can anybody tell me how to extend asterisk so it'll support H263++?

...[2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 34 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 103 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 99 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format H263 for ID 34 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format h263-1998 for ID 103 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format H264 for ID 99 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Capabilities: us - 0x3c0002 (gsm|h261|h263|h263...

...ion seems okay. When I leave a message, however, only the audio is recorded. Looking in the log file afterwards I see many messages like this: Mar 21 22:02:34 WARNING[2418] translate.c: No translator path from unknown to unknown Mar 21 22:02:34 WARNING[2418] file.c: Unable to translate to format h263, source format unknown For some reason, Asterisk seems to want to use h263, but that's not what is configured. Asterisk is also aware that the phone is using h261 because if I disallow that in sip.conf then video is stopped completely. After the voicemail message has been left, the normal au...

...SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833 host = dynamic qualify = yes allow = h263 video=yes videosupport=yes ; 112 is the X-Lite phone [112] type=friend host=dynamic user=112 username=112 secret=secret allow=all nat=no ------------- extensions.conf ------------- exten => 112,1,Dial(SIP/112) exten => 112,2,Playback(vm-nobodyavail) exten => 112,102,Playback(tt-allbusy)...

Hello list, what is the difference between these 2 codecs ? What codec to choose if bandwith is an issue ? (like in most cases I guess) Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101208/d611534a/attachment.htm

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4 (ulaw)): No such file or directory The file of course exists and it's chowned to asterisk.asterisk. I think it's a file format iss...

...ecify 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, preflan...

...ing from making this work to H.323 (which is a pain anyway) isn't working out so well. Thanks for any help, Rob >From sip.conf: [general] videosupport=yes [3005] type=friend secret=XXXXXX mailbox=3005 host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263 [3007] type=friend regexten=3007 mailbox=3007 secret=XXXXXX host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263

Hello, I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below: Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc)/video=0xc0000 (h261|h263), combined - 0x80000 (h263) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (te...

...ox. I did following configuration In Sip.conf videosupport=yes [phone1] type=friend host=dynamic context= employees mailbox=101 at default callerid="phone1<101>" disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] ?type=friend host=dynamic context= employees mailbox=102 at default callerid="phone2<102>" disallow=all allow=ilbc allow=g723 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263 allow=h263p allow=h261 In extensi...

hi, i'm a student of Udine University in Italy. I'm writing a project for Java Micro Edition to send a video stream from pc to phone. The player in that Java Platform doesn't work so I have to write "some" code. I'm looking for a java implementation for h263-mpg codec. Can you help me please? I'was looking for that since 10 days without results very glad for any information. bye -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora-dev/attachments/20061004/a78156a1/attachment.html

Hi Guys, Do you know of any asterisk community version that does video codec trans-coding or in other words supports video? I have 1.8.8.1 and I see h263.c format files but can't see that codec in make menuselect. it might be just a license issue (if h263 has to have license), but not sure if community versions offer video calls at all. Thank you! -- Khalid Touati -------------- next part -------------- An HTML attachment was scrubbed... URL:...

...ge deposited through voicemail() application. Everything works fine for the audio message if the format given in voicemail .conf is any audio format . e.g. in voicemail.conf give the format as format=gsm This case works fine. But if I give the following format in voicemail.conf format=h263|gsm And no video codec is sent from the phone, t hen the audio message is not stored in the database. I am getting the following warning: [Jan 13 15:14:17] VERBOSE[27656] app_voicemail.c:???? -- Recording was 0 seconds long but needs to be at least 1 - abandoning Although the nessage de...

...ssion c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:ON:WL:en-US:SI_SB_facebook:...

...s are quite simple, I use an asterisk with 4 peers all on the same LAN. My sip.conf looks like that: [general] bindport=1628 videosupport=yes limitonpeers = yes sendrpid = yes trustrpid = no disallow=all allow = gsm # 3 allow = alaw # 8 allow = ulaw # 0 allow = g722 # 9 allow = g726 # 111 allow = h263 # 34 allow = h263p # 98 allow = h264 # 99 [peerA] type=friend secret=kokolala nat=no host=dynamic canreinvite=no context=koko dtmfmode=rfc2833 disallow=all allow = gsm allow = alaw allow = ulaw allow = g722 allow = g726 allow = h263 allow = h263p allow = h264 qualify=no <snip more similar peer...

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

...recv m=video 48262 RTP/AVP 97 98 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:1920000 a=rtpmap:97 H264-RCDO/90000 a=fmtp:97 profile-level-id=008016;max- mbps=42000;max-fs=3600;max-smbps=323500 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500 a=rtpmap:99 H263-1998/90000 a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo a=rtpmap:34 H263/90000 a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200 a=rtpmap:31 H261/90000 a=fmtp:31 cif=1;qcif=1;maxbr=19200 a=rtcp-fb:* nack pli a=send...