search for: 3daudio

Hi, I have a few questions/remarks about the draft, in particular about the SDP usage of Speex (section 5): - Which samplerate should be used if you receive something like m=3Daudio 8008 RTP/AVP 97 a=3Drtpmap:97 speex/8000 a=3Dfmtp:97 sr=3D16000;ebw=3Dultra - There is an inconsistency in the description of the 'penh'parameter. According to the description, only the values '0' or '1' are valid. In section 6.1 however, the defaul...

...YE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY= ,ME SSAGE,REFER Content-Length: 226 Content-Type: application/sdp =20 v=3D0 o=3DHuaweiSoftX3000 228436 228436 IN IP4 189.8.113.170 s=3DSip Call c=3DIN IP4 189.8.113.170 t=3D0 0 m=3Daudio 49606 RTP/AVP 8 0 97 a=3Drtpmap:8 PCMA/8000 a=3Drtpmap:0 PCMU/8000 a=3Drtpmap:97 telephone-event/8000 a=3Dfmtp:97 0-15 =20 =20 and from Asterisk SIDE we have =20 =20 =20 12:54:30.364337 IP (tos 0x0, ttl 64, id 7173, offset 0, flags [none], proto: UDP (17)...

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

I'm a mathematician and programmer working on experimental surround sound techniques. Some of the ideas I'm working on require dozens of channels. These channels are often highly correlated and are very well suited to compression. I'm new to Vorbis. Does it attempt to address such issues? Thanks, --Richard --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project