search for: faststart

Hi, find enclosed a patch to implement faststart (or prebuffering). Basically, it stores a number of "refbuf"s, configurable by seconds. When a new client connects, it adds the prebuffers to que clients' queue so they are delivered as fast as permitted by the connection. As result, the players prebuffers are filled up faster gi...

Hi Mike, find the patch for faststart that takes in account different logical streams. I tried with ices' playlist, it works just fine [*]. I could be still further optimised, but it will make the code less clear (for example, I could check for has_paredata before checking for serailno, which saves a couple of calls and also wi...

On Sunday 19 January 2003 03:35, Michael Smith shaped the electrons to shout: > There's a bad interaction here between the predata and this faststart. > Suppose we have two logical streams (A and B), and the stream has just > recently got to logical stream B. Now, this means the fast start queue > contains the end of stream A, then the stream B headers, then the start > of stream B audio data. Good point, I didn't think about th...

On Sunday 19 January 2003 08:09, Ricardo Galli wrote: > Hi, > find enclosed a patch to implement faststart (or prebuffering). > > Basically, it stores a number of "refbuf"s, configurable by seconds. When > a new client connects, it adds the prebuffers to que clients' queue so > they are delivered as fast as permitted by the connection. As result, the > players prebuffers are...

On Monday 20 January 2003 06:16, Ricardo Galli wrote: > Hi Mike, > find the patch for faststart that takes in account different logical > streams. I tried with ices' playlist, it works just fine [*]. > > I could be still further optimised, but it will make the code less clear > (for example, I could check for has_paredata before checking for > serailno, which saves a couple...

One thing I noticed though when reading about ffmpeg to make the h.264 was that for internet, you could use qt-faststart to move some stuff (I think called MOOV) to the front of the file allowing the file to start playing before it was finished downloading. That certainly helps quite a bit, I was wondering if there is a similar type of thing for theora? Theora files I host do not seem to start playing until the...

...| V V [Asterisk]-----------(RTP)----------[Terminating gateway] | (Signalling + RTP) | (Zaptel/SIP/H323/SCCP phones) There are some requirements for me to connect to this switch: 1. I must support H245 tunneling and faststart (working fine) 2. I must dynamically negotiate the codecs (i.e. send multiple codecs as part of the faststart and the softswitch will decide which of the codecs to use based on the terminating gateway's capabilities). The codec picked will be passed back in the return faststart from the gateway...

...| V V [Asterisk]-----------(RTP)----------[Terminating gateway] | (Signalling + RTP) | (Zaptel/SIP/H323/SCCP phones) There are some requirements for me to connect to this switch: 1. I must support H245 tunneling and faststart (working fine) 2. I must dynamically negotiate the codecs (i.e. send multiple codecs as part of the faststart and the softswitch will decide which of the codecs to use based on the terminating gateway's capabilities). The codec picked will be passed back in the return faststart from the ga...

On Tuesday 21 January 2003 11:37, Ricardo Galli wrote: > Hi Mike > Find enclosed the patch with the fastart implementation for vorbis (for > the moment). > Now is based on size in bytes and the buffers are sent all together with > pre_data. > > Hope you like it. At least is the smallest one: :-) Ricardo, This is much improved. There are a few things to clean up (comments,

...iting > the generic implementation for get_predata() (which at the moment only > mp3 will use), but the design looks fine. I'm unlikely to have time to Hope you read this message before leaving the country... In order to implement a generic get_data function, which has to check for teh faststart_queue, and other fields, I must move the following fields: long faststart_size; /* If > 0, max bytes in queue */ refbuf_queue_t *faststart_queue; long faststart_queue_size; /* In bytes */ from the _state structure (which differs according to the format) to format_plugi...

Hi Mike Find enclosed the patch with the fastart implementation for vorbis (for the moment). Now is based on size in bytes and the buffers are sent all together with pre_data. Hope you like it. At least is the smallest one: :-) -rw-r--r-- 1 gallir gallir 5486 2003-01-21 01:29 update3c.diff -rw-r--r-- 1 gallir gallir 10014 2003-01-19 20:14 update3b.diff -rw-r--r--

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Sorry. The patch is hopely here. On Saturday 18 January 2003 03:38, Michael Smith shaped the electrons to shout: > > Hi, > > find a patch which is an update to a patch sent months ago. Before > > it was in net/sock.c, now I moved it to format.c, so net CVS module > > is not affected. It polls the socket before trying to

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want...

...fallthrough=yes [internal] exten => 1000,1,Dial(SIP/1000) exten => 1000,2,HangUp() exten => _XXXX,1,Dial(H323/${EXTEN}@Avaya) exten => _XXXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya) exten => _XXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya) *Here is also the content of my ooh323.conf:* [general] faststart=yes h245tunneling=yes gatekeeper=DISABLE bindaddr=10.1.129.231 port=1720 callerID="ALT Asterisk PBX" progress_setup=8 progress_alert=8 disallow=all allow=all dtmfmode=inband faststart=yes context=internal [Avaya] type=friend context=internal host=10.1.129.247 port=1720 canreinvite=no dis...

...ght be idiot...! ;) Please forgive me!) I succeed to make H323 call when ooh323c is configured as gateway (gatekeeper=DISABLE in ooh323.conf). When I put gatekeeper= ip_address, and add an account as follow : [aaa] type=friend username=aaa password=xxxx host=dynamic context=test incominglimit=4 faststart=yes h245Tunneling=yes , my H323 softphone can't register. ("sent GRQ"..."gatekeeper not responding") My questions are : 1/ Can ooh323c work as gatekeeper (if yes, even if it is installed on the same box as asterisk)? 2/ if yes, Do you know tutorials for doing this? o...

...323<->H323 is working, as well as SIP<->SIP. Running CVS version from yesterday. Used codecs are G711U & A, G723.1 and G729. If I just use G711 it's the same. SIP EP has to call first when * is started to make it work. Any hint? Also, H323 is still broken and working without FastStart. Is there a workaround existing? Regards -- Daniel

...er 17 cid 0x1289 The below is the configuration & logs, ------------------------------------------ Avaya :?AVAYA G650 S8800 Asterisk: 1.4.31 Ooh323 config: Objective Open H.323 Channel Driver's Config: IP:Port: 0.0.0.0:1720 H.225 port range: 12030-12230 FastStart yes Tunneling yes CallerId asterisk MediaWaitForConnect no Gatekeeper: No Gatekeeper H.323 LogFile: /var/log/asterisk/h323_log Context: default Capability: 0xf (g723|gsm|ulaw|alaw) DTMF Mode: rfc2833 AccountCode: ast_h32...

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

...keeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912 ID - HMA0200.10szxn-xxxx e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 gatekeeper=AVS@210.21.118.XXX gatekeeperTTL=600 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERI...

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web