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Transcoding between CUCM and CUCME

fjroblesp
Level 1
Level 1

Dear experts,
My objective is to have one conversation between 2 phones where one end (CUCM phone) is sending/receiving G.711, while the other end (CUCME SCCP phone) is sending/receiving G.722 or G.729.
The connection between CUCM and CUCME is an H.323 gateway configured on CUCM (I could not get G.722 to work on a SIP trunk).
I have the DSP resources registered, but the phones always negotiate to a common codec (either G.711, G.722 or G.729, whichever is specified under the dial-peer that is compatible with the codec configured inside Region_HQ).
dial-peer voice 100 voip
destination-pattern [15]...
session target ipv4:10.10.210.10
codec g722-64
BR2-RTR#sh sccp
SCCP Admin State: UP
Gateway Local Interface: Loopback0
        IPv4 Address: 10.10.110.3
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.10.210.10, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 1
                Trustpoint: N/A

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.10.210.10, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 4, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g723_6.3, Maximum Packetization Period: 60
Supported Codec: g723_6.3, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

BR2-RTR#
BR2-RTR#
BR2-RTR#
BR2-RTR#sh dspfarm all
Dspfarm Profile Configuration

Profile ID = 2, Service = TRANSCODING, Resource ID = 1 
Profile Description : 
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP   Status : ASSOCIATED
Resource Provider : FLEX_DSPRM   Status : UP
Number of Resource Configured : 2
Number of Resource Available : 2
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
Codec : g722-64, Maximum Packetization Period : 30
Codec : g729br8, Maximum Packetization Period : 60
Codec : g729r8, Maximum Packetization Period : 60
Codec : g723r53, Maximum Packetization Period : 60
Codec : g723r63, Maximum Packetization Period : 60
Codec : gsmamr-nb, Maximum Packetization Period : 60
Codec : ilbc, Maximum Packetization Period : 120
Codec : pass-through, Maximum Packetization Period : 0


SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   24.3.3   UP     N/A  FREE  xcode  1      -         -         -       
0    1   24.3.3   UP     N/A  FREE  xcode  1      -         -         -       

Total number of DSPFARM DSP channel(s) 2

BR2-RTR#
I have also configured IP-IP gateway:
voice service voip
allow-connections h323 to h323
Capturing the packets, I can see that the RTP flow is between CUCM Phone and CUCME router, and between CUCME router and CUCME Phone.
Could anybody please let me know if this is achievable and, if so, what I am missing?
Thank you very much for your help!!

1 Reply 1

Andrew Latamore
Level 1
Level 1

Hi Javier!  Looking at the issue you are encountering, I am wondering why you would not want the phones to negotiate a common codec.  This is the default behavior in order to avoid using transcoding resources (which involve costly DSPs).  At what point in your topology are you trying to use the transcoding?  You could force your ephone's to use a specific codec on your CME, then force your dial-peers to and from CUCM to use a different codec.  That would effectively force the use of transcoding, but I am still not sure what would be accomplished by doing this.  On the CUCM side remember that the region settings do not specifically hard code codecs, as they do in CME/gateways.  The region relationship settings allow CUCM to negotiate codecs down to the complexity chosen.  CUCM will attempt to seek the highest bandwidth codec within the range but will allow lower bandwidth codecs to be used.  For example if two regions are set to use G729, they will only be able to use G729 (nothing is lower).  However if they are set to G711 they can negotiate G711 and G729, but G711 will be preferred.  Therefore the best place to hard code your codecs would be on the gateway side of your configuration, because you have more options to hard set codec use.

So in the above mentioned idea of hard coding the ephones and dial-peers in the CME, you could force the use of transcoding.  However, remember that your voice quality on a given call will only be as good as the worst quality leg of that call.  So if you have one leg using a high quality codec and another using a low quality codec with transcoding inbetween your quality will only end up being perceived as being as good as the low quality leg.  I hope this helps a bit.

If this answers the question, or if the problem has already been resolved, please mark it as such.  That will avoid others on the forum from continuing to respond if it is no longer an issue for you.  Thanks!

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