[cisco-voip] CUBE not requesting codec... call fails... need to force SDP in invite...

[Nick Matthews]

Service providers don't like delayed offer.

You can do DO-EO conversion, but it requires SIP-SIP and won't work on
H.323-SIP.

For G.729:
You have to use a software mtp to do G729.  CUCM doesn't support G.729
in software MTP's, and the PVDM's won't do G.729-G.729, so it only
leaves you with router software MTPs.  If you haven't configured your
MTP in the MRGL for your H.323 gateway with a SW MTP with G.729
configured it won't work properly.

For G.711:
I would configure 'codec g711ulaw' on all dial peers, and drop the
voice-class codec.

If you can switch the H.323 gateway to a SIP trunk trunk, configured
DO-EO conversion:

voice service voip
 sip
  early-offer forced

-nick

On Tue, Jan 24, 2012 at 5:03 PM, Jonathan Charles <jonvoip at gmail.com> wrote:
> From the attached debug, we are sending an invite without codec info;
> carrier comes back and says G729 and we drop it. Users hear one ring and
> fast busy.
>
> Route is running SP Services 15.1.4M1
>
> Network is Phone to CCM, H.323 to CUBE, SIP to Provider.
>
> LD calls work cuz they are G.711 all the way, 800 numbers fail (cuz the
> carrier points it at a G.729 trunk).
>
>
> I have implemented the fix here:
>
> http://www.markholloway.com/blog/?p=1325
>
> But it doesn't work.
>
> The H.323 gateway does NOT have MTP required checked currently (it made no
> difference.
>
>
>
>
> [relevant config]
> !
> voice service voip
>  ip address trusted list
>   ipv4 10.0.0.0 255.0.0.0
>   ipv4 172.16.0.0 255.240.0.0
>   ipv4 172.20.0.0 255.255.255.0
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>  fax protocol pass-through g711ulaw
>  h323
>   h225 display-ie ccm-compatible
>  modem passthrough nse payload-type 101 codec g711ulaw
>  sip
>   bind control source-interface GigabitEthernet0/0
>   bind media source-interface GigabitEthernet0/0
> !
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g729r8
> !
> dial-peer voice 201 voip
>  translation-profile outgoing Last10
>  preference 1
>  destination-pattern ^1[1-9].........
>  session protocol sipv2
>  session target ipv4:72.11.192.82
>  session transport udp
>  voice-class codec 1
>  no voice-class sip pass-thru content sdp
>  dtmf-relay rtp-nte
>  no vad
> !
> dial-peer voice 202 voip
>  translation-profile outgoing Last10
>  preference 2
>  destination-pattern ^1[1-9].........
>  session protocol sipv2
>  session target ipv4:72.11.193.82
>  session transport udp
>  no voice-class sip g729 annexb-all
>  no voice-class sip pass-thru content sdp
>  dtmf-relay rtp-nte
>  no vad
> !
>
>
> So, the question is, how do I force an SDP specifying G.711 in the invite?
>
>
>
>
> Jonathan
>
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