asterisk users - May 2006 - SIP to IAX - forcing codec pass thru

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@customeripaddress/441376350002") in new stack
     -- Called user:3l3phant@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

-- 
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
  -- engineering & hosting services for email, web and voip --
   -- http://www.peter.me.uk/  -- http://www.voip.org.uk/ --

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@customeripaddress/441376350002") in new stack
>      -- Called user:3l3phant@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
>