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Thread: Re: AMR in H324M<->SIP
Re: AMR in H324M<->SIP
country flaguser name
Spain		 2007-07-18 09:46:26 


---------- Original Message
----------------------------------
From: Klaus Darilion <klaus.mailinglistspernau.at>
Reply-To: Development discussion of video media support in
Asterisk<asterisk-videolists.digium.com>
Date:  Wed, 18 Jul 2007 14:06:56 +0200

>
>
>Sergio Garcia Murillo wrote:
>> Hi everyone,
>> 
>> Paul Bagyenda sent me a modification of his
amr_codec to support octect
>> aligned mode in if2 (the format needed for h324m).
>> I haven't not been able to test it (I don't have an
isdn avaiable), but it
>> would be great if any of you make it works.
>> 
>> In codecs.conf, add:
>> 
>> [amr]
>> octet-aligned=1
>
>With octet-aligned=1 Asterisk fails to load:
>
>codec_amr: enc_mode = 7, dtx = 0
>   == Registered translator 'amrtolin' from format amr
to slin, cost 2
>asterisk: codec_amr.c:351: lintoamr_frameout: Assertion
`"Padding bits 
>cannot be > 0 in octet aligned mode! "
&& 0' failed.
>Aborted
>


As I said, the patch is not mine, I'll try to contact Paul
to see if he can
solve it.

Best regards
Sergio 

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Re: AMR in H324M<->SIP
country flaguser name
Austria		 2007-07-18 10:14:23 

Sergio Garcia wrote:
> As I said, the patch is not mine, I'll try to contact
Paul to see if he can
> solve it.
> 


Hi Sergio!

Have you ever managed to gateway a call from H324M to a SIP
client? If 
yes - did you used AMR at the SIP client or did Asterisk
transcoding? If 
asterisk did transcoding - which AMR codec do you used? If
Asterisk did 
not transcoding - which SIP client have you used?

thanks
klaus

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Re: AMR in H324M<->SIP
user name
 2007-07-19 04:15:41 

  


  

    HI all,

I9;m having the same problem here. Need to interconnect simple SIP phones (soft or hard) with H324M calls.
So I either need a SIP phone that speaks AMR, or asterisk needs to transcode.
Have tried to implement the amr codec as described by Paul (
http://lists.digium.com/pipermail/asterisk-dev/2007-March/026446.html)
All compiles well, but I don't get any audio throughput.


Has anybody had more luck with this?

For the record, I know there's a licensing issue with AMR.
But since VoiceAge is giving away free amr codec samples (
http://www.voiceage.com/freecodecs.php) for development purpose,
I guess we can try develop it into Asterisk...

Koen

On 7/18/07, Klaus Darilion
 < klaus.mailinglistspernau.at">klaus.mailinglistspernau.at> wrote:



Sergio Garcia wrote:
>; As I said, the patch is not mine, I'll try to contact Paul to see if he can
> solve it.
>


Hi Sergio!

Have you ever managed to gateway a call from H324M to a SIP client? If

yes - did you used AMR at the SIP client or did Asterisk transcoding? If
asterisk did transcoding - which AMR codec do you used? If Asterisk did
not transcoding - which SIP client have you used?

thanks

klaus

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