[Asterisk-video] AMR passthrough patch on bugtracker

Klaus Darilion klaus.mailinglists at pernau.at
Thu Oct 18 08:21:47 CDT 2007


Hi Koen!

I just tested the AMR codec again in the following scenario:

                 +----+         +----+
xlite <--ulaw-->|    |<--AMR-->|    |
                 |Ast |         |Ast |
PSTN  <--ISDN-->|eri |<--AMR-->|eri |
                 |sk 1|         |sk 2|
                 +----+         +----+

Asterisk 1 is SVN-branch-1.4-r75450M with AMR patch from sip.fontventa.com

Asterisk 2 is SVN-trunk-r82546M with AMR passthrough patch from 
bugs.digium.com

I verified with wireshark: There is really AMR used between the 2 
Asterisk servers. Thus Asterisk 1 doing transcoding between SIP-SIP and 
SIP-ISDN.

Maybe there is a bug in the MAR codec which is only triggered in your 
scenario or happens only with newer Asterisk versions.

regards
klaus


Koen Van Impe schrieb:
> Hi Klaus,
> 
> I tried your patch that uses the amr codec.
> Managed to build it with Asterisk 1.4.13.
> We have a 3G gateway that talks SIP to this Asterisk and on the other 
> side a plain X-Lite that registers with this Asterisk (using g711 alaw).
> 
> I can't call X-Lite. Error: "chan_sip.c:2963 sip_call: No audio format 
> found to offer"
> However, the amr codec seems to be available:
> 
>         INT    BINARY        HEX   TYPE       NAME   DESC
> --------------------------------------------------------------------------------
>           1 (1 <<  0)      (0x1)  audio       g723   (G.723.1)
>           2 (1 <<  1)      (0x2)  audio        gsm   (GSM)
>           4 (1 <<  2)      (0x4)  audio       ulaw   (G.711 u-law)
>           8 (1 <<  3)      (0x8)  audio       alaw   (G.711 A-law)
>          16 (1 <<  4)     (0x10)  audio   g726aal2   (G.726 AAL2)
>          32 (1 <<  5)     (0x20)  audio      adpcm   (ADPCM)
>          64 (1 <<  6)     (0x40)  audio       slin   (16 bit Signed 
> Linear PCM)
>         128 (1 <<  7)     (0x80)  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)
>        8192 (1 << 13)   (0x2000)  audio        amr   (AMR NB)
>       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 << 19)  (0x80000)  video       h263   (H.263 Video)
>     1048576 (1 << 20) (0x100000)  video      h263p   (H.263+ Video)
>     2097152 (1 << 21) (0x200000)  video       h264   (H.264 Video)
> 
> On the bright side: amr works fine in pass-through. So it is recognized 
> correctly during call setup, right?
> 
> Regards,
> 
> Koen
> 
> 
> 
> On 9/17/07, *Klaus Darilion* <klaus.mailinglists at pernau.at 
> <mailto:klaus.mailinglists at pernau.at>> wrote:
> 
>     fyi: http://bugs.digium.com/view.php?id=10741
>     <http://bugs.digium.com/view.php?id=10741>
> 
>     feedback welcome
> 
>     regards
>     klaus
> 
>     _______________________________________________
>     --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
>     asterisk-video mailing list
>     To UNSUBSCRIBE or update options visit:
>        http://lists.digium.com/mailman/listinfo/asterisk-video
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video



More information about the asterisk-video mailing list