search for: rfc3551

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

...very good. However, things are not so easy with Eyebeam and the IP650. When a call is placed between those two the audio stream from Eyebeam to the IP650 is never heard. The audio from the IP650 to Eyebeam is heard, and very good quality. David Frankel of ZipDX tells me that there is an error in RFC3551 such that G.722 RTP clock/timestamps are actually wrong. To quote the RFC directly. "Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged...

...is the output of the encoder? Also the config value as given in the RFC6716, 16...19 | CELT-only | NB | 2.5, 5, 10, 20 ms 16 corresponds to 2.5 ms 17 corresponds to 5 ms 18 corresponds to 10 ms 19 corresponds to 20 ms Is this correct representation of the data? Also in the RFC3551 the payload values are specified but does not give any information on whether OPUS encoder has been used? Regards Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20160509/24b6856d/attachment.html>

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

...(0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 16384 (1 << 14) (0x4000) audio siren14 (ITU G.722.1...

...> 512 (1 << 9) (0x200) audio speex >> (SpeeX) >> 1024 (1 << 10) (0x400) audio ilbc >> (iLBC) >> 2048 (1 << 11) (0x800) audio g726 >> (G.726 RFC3551) >> 4096 (1 << 12) (0x1000) audio g722 >> (G722) >> 8192 (1 << 13) (0x2000) audio siren7 (ITU >> G.722.1 (Siren7, licensed from Polycom)) >> 16384 (1 << 14)...

...x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 65536 (1 << 16) (0x10000) image jpeg (JPEG image) 131072 (1 << 17) (0x20000) image png (PNG image) 262144 (1 << 18) (0x40000) video h261 (H.261 Video) 524288 (1 &...

...x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 65536 (1 << 16) (0x10000) image jpeg (JPEG image) 131072 (1 << 17) (0x20000) image png (PNG image) 262144 (1 << 18) (0x40000) video h261 (H.261 Video) 524288 (1 &...

Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware

...x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 65536 (1 << 16) (0x10000) image jpeg (JPEG image) 131072 (1 << 17) (0x20000) image png (PNG image) 262144 (1 << 18) (0x40000) video h261 (H.261 Video) 524288 (1 &...

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

...(0x200) audio speex >>>> (SpeeX) >>>> 1024 (1 << 10) (0x400) audio ilbc >>>> (iLBC) >>>> 2048 (1 << 11) (0x800) audio g726 >>>> (G.726 RFC3551) >>>> 4096 (1 << 12) (0x1000) audio g722 >>>> (G722) >>>> 8192 (1 << 13) (0x2000) audio siren7 >>>> (ITU >>>> G.722.1 (Siren7, licensed from Polycom)) >&g...

...g729 > (G.729A) > 512 (1 << 9) (0x200) audio speex > (SpeeX) > 1024 (1 << 10) (0x400) audio ilbc > (iLBC) > 2048 (1 << 11) (0x800) audio g726 > (G.726 RFC3551) > 4096 (1 << 12) (0x1000) audio g722 > (G722) > 8192 (1 << 13) (0x2000) audio siren7 > (ITU G.722.1 (Siren7, licensed from Polycom)) > 16384 (1 << 14) (0x4000) audio sir...