[Asterisk-Dev] SIP DTMF causing voice quality loss?

[John Todd]

>I've been playing with SIP a lot lately and we rolled out some more 
>of our employees on SIP based long distance (testing to see if it's 
>feasible to collocate our Asterisk at our data center with a PRI - 
>hence the talks before about sharing PRI, etc.)..
>
>
>
>I'm seeing an odd issue, I'm testing through IConnectHere with 
>limited success - however I seem to have one thing that may be 
>reproducible (too early to say if reproducible):
>
>             Layout:  POTS based phone, hooked to Zhone Channel bank 
>- Zhone to Quad T1 Digium Card on Asterisk server.  Asterisk then 
>Dial/SIP to analog based phone system. 
>
>Trying: Analog phone system (called party) says press 123 for 
>sales.. We press 123.  Suddenly call quality is horrible (we can 
>hear fine about 75% of the time, other 25% no audio, called party 
>says can't hear us hardly) - other DTMF tones don't seem to go 
>through right (maybe unclear so not being recognized).

No idea on this one.  What is your dtmf setting on the SIP peer?

>Works: Calling what I would guess is another digital PBX works fine 
>using this same layout.  I've tried a few banks / large businesses 
>and no problems.
>
>Works: Can call a phone via SIP and recognize DTMF from called party 
>(i.e. # to transfer)

Still no "T" option working that I can determine, though.  :(  Not 
that major an issue, but still problematic for when I want to send 
someone to my IVR or send them to a DISA line after calling them.

>
>
>Any ideas?

Change dtmfmode= in your sip.conf file for that peer to either 
rfc2833 or inband.  I have yet to get iconnecthere working with DTMF, 
and though some people here on the list say they've had it work, I 
have not seen any configs, nor will they comment on how they did it.

>
>We should have another SIP based gateway up next week and we now 
>have pricing on DC based PRI that's good so we may just go with PRI. 
>I'd rather avoid the cross connect fees / loops on the PRI since we 
>can get SIP based easier and have plenty of bandwidth - it seems 
>like just another thing to go wrong in Asterisk using PRI instead of 
>Cisco delivered SIP (maybe I'm backwards on this).

I think I know the network configurations you're talking about.  :)

By using a SIP provider, you lose some bandwidth advantages that IAX 
can provide to you (if you want, or care.)  However, you are correct 
in that you don't have to pay for a PRI into your facility and then 
connect it to your gear.  If you can get SIP directly via IP (across 
the Internet) is the "best" solution as long as 
bandwidth/jitter/latency/cost of bits are acceptable for your 
circumstances.  The second best path would be to find a provider who 
can co-locate an Asterisk box and give you zero-mile local loop costs 
for a PRI into their switch; the cost for a 1u box plus some 
bandwidth for IAX communications is almost always cheaper than the 
delta of the cost for that same PRI being dragged through the RBOC.

>  PS: How does IAX handle DTMF between Asterisk servers?

Works fine.  I use it all the time.  SIP -> * -> IAX -> * -> PRI 
works like a charm, though the tones are a bit short by default. 
I've never had problems with any of the IVR systems I've used, though.

JT