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

[Jonathan Charles]

Thanks, that fixed it...

Along with changing the H.323 gateway to outbound fast start...

Now, interestingly, you can only specify one codec for the software MTP...
 but it doesn't seem to matter, both 711 and 729 calls work fine...




Jonathan



On Tue, Jan 24, 2012 at 4:44 PM, Nick Matthews <matthnick at gmail.com> wrote:

> 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
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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