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CM to SM to CM G.722-64K Codec SIP Trunk Calls

CM to SM to CM G.722-64K Codec SIP Trunk Calls

CM to SM to CM G.722-64K Codec SIP Trunk Calls

(OP)
I have a situation where I can NOT get a CM to CM call to negotiate at the G.722-64K Codec within the same telephony network.
Upon investigating, I am informed that if going across the WAN, regardless what your internal codec is set to, the carrier decides the best default codec for the call.
Has anyone else come across this problem?

Here's what I have:
CM A:
Location: NJ
Phone Model: 9621
CM Release: 7.1
Network Region: 5
Codec Set: 5
1: G.722-64K
2: G.711MU
3: G.729
4: G.711A
5: G.729A
Intra-region IP-IP Direct Audio: Y
Trunk Group: 100
Signaling Group: 100
Direct IP-IP Audio Connections: Y
IP Audio Hairpinning: N

Inter Network Region Connection Management to NR 100: Codec Set 2
Intervening NR: 100
Codec Set 2
1: G.722-64K
2: G.711M


CM B:
Location: Zurich
Phone Model: 9621
CM Release: 7.1
Network Region: 1
Codec Set: 1
1: G.722-64K
2: G.711A
3: G.711MU
4: G.729
5:
Intra-region IP-IP Direct Audio: Y
Trunk Group: 10
Signaling Group: 10
Direct IP-IP Audio Connections: Y
IP Audio Hairpinning: N

Inter Network Region Connection Management to NR 4: Codec Set 1
Intervening NR: 4
Codec Set: 1
1: G.722-64K
2: G.711A
3: G.711MU
4: G.729
5:

Guillermo "Will" Ramirez
Avaya Telephony & Unified Communications Professional

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

What's your SIP signaling group look like between CM-A and B?

Based on this snippet from the 9600 H323 admin guide below (presuming you're h323 phones and SIP trunks between CMs...), I think you'll want to check all the early media stuff in your sig groups. There's early H323 station media and early direct IP media - settings that will make CM's initial invite out contain as media IP the IP of the endpoint making the call rather than a gateway first that it will try to shuffle off of once the call is established. If your initial offer includes a media gateway IP, then it'll never include G722, and likely won't shuffle up from 711 thru a DSP to 722 direct once established.

Also, check the network region of the far end of whatever sig group you have from CM A-->B and vice versa. Specifically that far end network region in a sig group is what CM uses to decide the codecs to offer from the network region of the source making the call . Say your phone is in NR2 and the far end node of the sig group is NR20 and in your NR connectivity you have regions 2-->20 using a codec set that might not be 722 first. Stuff like that.

CODE -->

Enable Wideband Audio, by using the Change IP
codec command on CM. Ensure that G.722–64K is
first on the list of codecs. Note that wide band audio
works only for direct-IP calls between two 96xx
endpoints, either with both registered to the same
server, or registered to different servers when
connected by IP trunks. Calls between two 96xx
phones connected by an IP trunk do not currently
support wide band audio when the call is shuffled
so that the media travels directly between the two
96xx IP phones. Calls that involve three or more
parties, even if all parties use 96xx IP phones, do
not use wide band. Calls between two 96xx IP
phones where audio is terminated at a port network/
gateway (PN/GW) media resource will not use
wideband.
Ensure that G.722 is added to all codec-sets that
can possibly be used between all regions on the IPNetwork Regions form where 96xx IP phones exist.
Technically, G722 does not need to be first. What is
needed, however, is that all the non media
processor-supported codecs (G722, SIREN, etc.)
be placed before the media processor-supported
codecs (G711, G729, G726, G723) 

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

(OP)
Thank you Kyle. I will re-confirm.

Guillermo "Will" Ramirez
Avaya Telephony & Unified Communications Professional

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

(OP)
OK so based on direction from kyle555 and additional research, we now have this working properly. Yippee!

The Avaya documentation is a bit deceiving, but even with NE NR & FE NR shuffling turned on in order to utilize the DSP resources on the G450 gateways as follows:
Intra-Region IP-IP Direct Audio: Yes
Intra-Region IP-IP Direct Audio: Yes
IP Audio Hairpinning: N

A change to the Trunk Group's Signaling Group needs to be applied as recommended and as follows:
Direct IP-IP Audio Connections?: Y
Initial IP-IP Direct Media?: Y

We now have successful calls on the G.722-64K codec between different locations on different CMs within our Telephony Network and over SIP Trunks.



Guillermo "Will" Ramirez
Avaya Telephony & Unified Communications Professional

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

awesome!

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

@gwramirez - thank you for putting your fix/resolution up here publicly. It's going to help others looking for your problem in the future. I think this may also help me in a lab scenario I was having also. I've purple starred it.

RE: CM to SM to CM G.722-64K Codec SIP Trunk Calls

(OP)
Thx Randy!

God Bless...

Guillermo "Will" Ramirez
Avaya Telephony & Unified Communications Professional

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