[Asterisk-Users] SIP to IAX - forcing codec pass thru
bails
bails at westcomuk.com
Mon May 22 04:14:31 MST 2006
Hi Peter, as one of your customers I would ask you not to dissallow g729
on IAX2 as we currently use it extensively.
Bails
Mark Phillips wrote:
> Hi Peter,
>
> I don't see any codec allow=blah statements. If your end user has
> something like
>
> [gradwell]
> disallow=all
> allow=gsm
>
> Then you'll be forced to send them a GSM coded call.
>
> Why not force the codec at your end by only supporting one? If the
> customer then transcodes the call when it gets forwarded to his handset
> there's not much you can do about that but at least you'll have handed
> the call off in the best format you can source.
>
> Mark
>
> On Mon, 2006-05-22 at 09:57 +0100, Peter Gradwell wrote:
>
>>hi
>>
>>We take calls inbound via SIP from our Cisco PSTN gateways, and pass it
>>to customers using IAX (they run their own asterisk servers).
>>
>>We've noticed that asterisk is transcoding the call into a different
>>codec, if the customer prefers a codec different to that which our cisco
>>gw prefers. As such, the quality of the call can degrade.
>>
>>We'd rather asterisk just passed through the RTP stream and maintained
>>the same codec, so that all asterisk did was signalling conversion.
>>
>>
>>
>>sip.conf...
>>
>>---
>>
>>[sip-router-1.gradwell.net]
>>context=sip-inbound
>>type=peer
>>host=sip-router-1.gradwell.net
>>
>>[sip-router-2.gradwell.net]
>>context=sip-inbound
>>type=peer
>>host=sip-router-2.gradwell.net
>>
>>---
>>
>>iax.conf...
>>
>>[general]
>>bandwidth=high
>>disallow=lpc10
>>jitterbuffer=yes
>>dropcount=2
>>maxjitterbuffer=500
>>maxexcessbuffer=80
>>minexcessbuffer=10
>>jittershrinkrate=1
>>tos=lowdelay
>>
>>
>>---
>>
>>when a call comes in, we dial something like this, in our dial plan:
>>
>> -- Executing Goto("SIP/213.166.5.134-118f5310",
>>"sip-users|7770002|1") in new stack
>> -- Goto (sip-users,7770002,1)
>> -- Executing Dial("SIP/213.166.5.134-118f5310",
>>"IAX2/user:pass at customeripaddress/441376350002") in new stack
>> -- Called user:3l3phant at customeripaddress/441376350002
>> -- Call accepted by customerip (format alaw)
>> -- Format for call is alaw
>> -- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310
>>
>>thanks
>>peter
>>
>
>
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