Issue on gateway 2811 as trascoding.

[Pablo Roman Reinoso]

Hi everybody

IPPhones(sccp)-CCM(trunksip) <-- || sip-gw2811(transcoding)-sip || <-- CallCenter-IPPhones(sip) <-- (PSTN)

I have a gateway 2801(transcoding) connected to a CallCenter "Interactice Intelligent" by sip. Callcenter only use g711 on yours sipphones and over the other side there is

a CCM 4.2.3 that use g729 on sccpphones.

The transcoding PDVM-2 is registred in CCM, when a incoming call from callcenter in to a sccpphone the call is connected and the gw2811 do transcoding but

when the user press hold and then resume only have one way audio from sccpphone to sipphone. For any reason don't recieve Rx Audio from gateway transcoding.

Any idea why i have one way audio when press hold and then return the call?

[g729 sccpphone ] <-- || g729 gw2811 g711 || <--[ g711 sipphone] <-- (PSTN)

something config on Callmanager.

region default serena

default g711 g729

serena g729 g711

sccpphone - trunksip - moh - transcoding

(serena) (default) (default) (default)

regards,

Pablo

[Chad Stachowicz]

pablo,

I have some bad news... There is no trancoding features available for sip to sip. You are going to have to use the same codec all the way through! Sorry, I know it sucks, big gotcha I found in CCIE Study!

HTH,

Chad

[Pablo Roman Reinoso]

Hi chad

Do you any document? because the transcoding work 50% the call is established on g729 to sccpphone and g711 to sipphone.

i say 50% because for any reason when at one established call if i put on hold and then resume have one way audio.

regards,

pablo

[paolo bevilacqua]

Actually I don't believe that transcoding for sip to sip is not supported.

Which exact IOS are you running on the GW doing transcoding ?

[Pablo Roman Reinoso]

i was used 124-15.T and updated to 124-15.T5 last version.

regards,

pablo

[paolo bevilacqua]

I'm hesitant in suggesting any upgrade in this case and think you should have the tac to reproduce problem in the lab, to be investigated and fixed.

[Chad Stachowicz]

Paolo,

I stand corrected at least per...

http://www.cisco.com/en/US/docs/ios/12_4t/12_4t4/ipip_sip.html

SIP-to-SIP protocol interworking capabilities of the Cisco Multiservice IP-to-IP Gateway support the following:

•Basic voice calls (G.711 and G.729 codecs)

•Calling/called name and number

•Codec transcoding (G.711-G.729)

I certainly have never gotten it to work though in my lab. I am going to give it another shot this week and post my results..

Chad

[Chad Stachowicz]

Just looked at your config again. The use of voice class codec will not allow for transcoding. Hardcode the dial-peers with both codec preferenced 1 and 2... This should allow transcoding to invoke...

try

dial-peer voice 10 voip

preference 1

tone ringback alert-no-PI

destination-pattern 8[5-6]..

codec g711alaw

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

dial-peer voice 11 voip

preference 2

tone ringback alert-no-PI

destination-pattern 8[5-6]..

codec g711ulaw

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

dial-peer voice 12 voip

preference 3

tone ringback alert-no-PI

destination-pattern 8[5-6]..

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

dial-peer voice 20 voip

preference 1

tone ringback alert-no-PI

destination-pattern [1-79]...

codec g711alaw

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

dial-peer voice 21 voip

preference 2

tone ringback alert-no-PI

destination-pattern [1-79]...

codec g711ulaw

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

dial-peer voice 22 voip

preference 3

tone ringback alert-no-PI

destination-pattern [1-79]...

session protocol sipv2

session target ipv4:10.1.24.208

dtmf-relay sip-notify rtp-nte

!

HTH,

Chad

[michaethompson]

why is it that the voice class command won't allow the XCoder to be invoked? wouldn't it depend on the supported codec in each voice class?

[Pablo Roman Reinoso]

i've probed forcing the codec but i can't recieve any calls from transcoding if i do that.

question:

The trunksip sip on CCM only work on g711, So Is it possible to have g729 over the ipphones with trasncoding if the call in by trunksip?

ip phones(g729)<-- CCM trunksip(g711) <-- sip gw transcoding

regards

Pablo

[Chad Stachowicz]

I agree, by all means it should, and in newer IOS this MAY have been fixed. In the voice-lab IOS it doesnt work... And its because its trying to negotiate it rather then knowing it needs to transcode. I think its by design..

Chad

[mshadbol]

TAC recently pointed me to this document about CUBE, and it states that SIP-to-SIP transcoding is not supported at 12.4(15)T and below, but doesn't suggest that it's supported later (like in the 'XY' release).

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1309838

(I have another question in this same topic in this months' CUBE special forum with the Product Manager - awaiting a response).

Furthermore, "voice class codec" can't be used on dial-peers in an IP-to-IP call leg. Automatic negotiation isn't supported, but 'hard-coded' codecs on dial-peers are. I don't have documentation supporting this, but was advised of this fact by the CUBE Product Manager.

Hope this helps,

-Martin