H.323 Outbound Fast Start Drop Calls from G.729 Region

Unanswered Question
Dec 14th, 2007


I have an IPIPGW with H.323 to a Cisco Callmanager and a SIP dial-peer to a Service Provider. The Service Provider only accespts SIP Invite messages with SDP according to the SIP standard, so the "outbound Fast Start" with G.729 codec and the "require MTP" check box are configured on the CCM.

I have transcoder running on my IPIPGW/VGW. Calls coming from IP Phones in a G.711 Region can be routed through the SIP provider or the PSTN with no problems. Transcoding from G.711 to G.729 through the SIP provider is working fine.

However, if I have an IP Phone in a G.729 region, those phones are not being able to complete any calls to the PSTN or the SIP provider. Calls disconnect immediately once the called party rings.

Any ideas?

I have this problem too.
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jawad1979 Fri, 12/14/2007 - 05:57


Here is the related config from the IPIPGW.

voice class codec 1

codec preference 3 g729r8 bytes 60

codec preference 4 g729br8 bytes 60

codec preference 5 g711ulaw

codec preference 6 g711alaw


dial-peer voice 600 voip CallManager Dial-peer

destination-pattern 6..

voice-class codec 1

voice-class h323 1

session target ipv4:

dtmf-relay h245-alphanumeric


dial-peer voice 99500 voip General Incoming Dial-peer

voice-class codec 1

incoming called-number .

dtmf-relay rtp-nte h245-alphanumeric

no vad

dial-peer voice 33333 voip SIP Provider dial-peer


destination-pattern 33.T

redirect ip2ip

voice-class codec 1

session protocol sipv2

session target ipv4:X.X.X.X

dtmf-relay rtp-nte digit-drop h245-alphanumeric

jhinton Fri, 12/14/2007 - 06:16

I think you need two trunks to the IPIPGW to acheive this. Utilize a SIP trunk for G.711 calls and an H.323 trunk for G.729 calls (both with MTP). I think the issue lies with having to specifically define a codec for call setup when using outbound fast start.

jawad1979 Fri, 12/14/2007 - 07:05


Thanks for your post, but transcoding from G.711 to G.729 over a SIP trunk to an IPIPGW is not yet supported with the latest IPIPGW IOS.

I compared two debugs from two phones in two different regions (G.711 and G.729).

I found out that the failed call has the following:

*Dec 14 14:59:07.371: //2629/xxxxxxxxxxxx/CCAPI/ccCallModify:

Nominator=0x18E00, Params=0x45417A70, Call Id=2629

*Dec 14 14:59:07.375: //2629/xxxxxxxxxxxx/CCAPI/cc_api_call_modify_done:

Result=0, Interface=0x44CE7904, Call Id=2629

*Dec 14 14:59:07.415: //2629/xxxxxxxxxxxx/CCAPI/ccCallModify:

Nominator=0x800, Params=0x45417A90, Call Id=2629

*Dec 14 14:59:07.415: //2630/xxxxxxxxxxxx/CCAPI/ccConferenceDestroy:

Conference Id=0x617, Tag=0x0

While the successful call has the following on the same sequence of the failed call:

*Dec 14 15:16:50.219: //-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:

Interface=0x44CE7904, Interface Type=9, Destination=, Mode=0x0,

Call Params(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,

Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

*Dec 14 15:16:50.219: //2647/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:

SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1

*Dec 14 15:16:50.219: //2647/xxxxxxxxxxxx/CCAPI/ccCallSetContext:


*Dec 14 15:16:50.223: //2647/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:



jhinton Fri, 12/14/2007 - 08:12

What do you get on a "debug ccsip messages" when you have a failed call?

jawad1979 Fri, 12/14/2007 - 10:49


I have partially solved the problem, I defined an incoming voip dial-peer for the CCM with a "codec transparent", IP Phones in a G.729 region can call the PSTN with no problems. Seems there is a general non-documented rule that whenever an "outbound Fast start" configured, the IPIPGW must have an incoming dial-peer with transparent codec.

However, those phones still not being able to make calls through the SIP provider, I tried the "codec transparent" trick for it, but still did not work.




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