08-25-2007 04:54 AM
Hi, I'm looking for a basic config for a 3725 with NM-HDV + PVDM to do SIP transcoding from g711 to g729. Anyone with some pointer where to look to start?
My setup is:
IPIVR<--SIP,G711-->3725<--SIP,G729-->Carrier
I have IOS image:
c3725-adventerprisek9_ivs-mz.124-16.bin
Most of the config I found is for Cisco CallMgr and SCCP. Is there config just for plain generic transcoding with SIP?
Any pointer appreciated.
Thanks in advance.
Here is my current config:
!
! No configuration change since last restart
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname A-37-01
!
boot-start-marker
boot-end-marker
!
aaa new-model
!
!
aaa authentication login local-auth local
!
aaa session-id common
clock timezone GMT 8
no network-clock-participate slot 1
no network-clock-participate slot 2
voice-card 1
dspfarm
dsp services dspfarm
!
voice-card 2
dspfarm
dsp services dspfarm
!
ip cef
!
!
!
!
ip name-server 192.168.1.1
!
!
!
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice class codec 2
codec preference 1 g729br8
!
voice class codec 3
codec preference 1 g729r8
codec preference 2 g723r53
codec preference 3 g723r63
codec preference 4 g723ar53
codec preference 5 g723ar63
!
!
!
!
!
!
!
!
!
!
!
!
ip ssh authentication-retries 2
ip ssh version 2
!
!
!
!
!
interface FastEthernet0/0
ip address 192.168.1.101 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
speed 100
full-duplex
!
interface FastEthernet0/1
no ip address
no ip redirects
no ip unreachables
no ip proxy-arp
shutdown
speed 100
full-duplex
!
router rip
network 192.168.1.0
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
ip http server
no ip http secure-server
!
no cdp run
!
!
!
!
control-plane
!
!
!
!
!
!
dspfarm transcoder maximum sessions 120
dspfarm rtp timeout 60
dspfarm connection interval 60
dspfarm
!
!
dial-peer voice 100 voip
tone ringback alert-no-PI
description Incoming
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte h245-alphanumeric
!
dial-peer voice 200 voip
description Outgoing
destination-pattern .T
voice-class codec 2
session protocol sipv2
session target ipv4:192.168.1.111
dtmf-relay rtp-nte h245-signal h245-alphanumeric
!
gateway
timer receive-rtcp 5
timer receive-rtp 180
!
sip-ua
retry invite 2
retry response 2
retry bye 2
retry cancel 2
!
!
!
!
gatekeeper
shutdown
!
!
line con 0
transport preferred none
line aux 0
exec-timeout 5 0
login authentication local-auth
transport preferred none
line vty 0 5
access-class 1 in
exec-timeout 5 0
login authentication local-auth
transport preferred none
transport input ssh
!
end
09-03-2007 01:49 PM
Transcoding doesn't care about the VoIP protocol, it's only for codec negotiation.
Did you ensure your incoming call is using the right dial-peer (show call active/history voice brief)?
Did you hear ring back on phone A, did the phone B ring out?
The call is dropped after you answer the call on phones B?
This documentation can help you...
http://www.cisco.com/en/US/products/ps6441/products_configuration_guide_chapter09186a0080541bf3.html
09-03-2007 05:00 PM
Yes, the call dropped with SIP error 480 (TIMEOUT) after I answered it. The codec I try to use is:
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
09-18-2007 01:13 PM
A transcoder will need to register to something- it uses SCCP and this is Master/Slave.
You should define start the telephony-service on the router and register the transcoder locally to the telephony-service on the router (you are not really going to setup the router to become a CME- this is just for the transcoder).
I think the bit of config you are missing is:
sccp
sccp local Vlanxxx
sccp ccm 1.1.1.1 priority 1
!
telephony-service
max-ephones 1
max-dn 1
ip source-address 1.1.1.1
sdspfarm units 1
sdspfarm transcode sessions x
sdspfarm tag 1 mtp
In your dial-peers do not use the voice class codec command otherwise you will not be able to use the transcoder- hardcode the codec to g729r8/g711ulaw...
One more thing- unless its a new feature, transcoding is not supported SIP-SIP. So you may have to use H323 on one leg of the call. Thats how it is in earlier 12.4 releases.
09-18-2007 03:19 PM
That is good advice by vmalhi. However in latest code it should be possible to use transcoding for everything without configuring sccp. See:
http://cisco.com/en/US/products/ps6706/products_feature_guide09186a008076161a.html
Remember to use an Ip-to-ip gw image which filename ends with _isv.
Hope this helps, please rate post if it does!
09-18-2007 03:46 PM
The transcoder needs to register to something- it can't invoke itself. You always need to associate it to you SCCP application that you define on the router and register it to one of CME/SRST/CCM. The document also states this, so I am not sure what you are saying?
09-18-2007 03:48 PM
Point taken and I stand corrected. Thanks!
09-18-2007 03:50 PM
No probs! You had me questioning myself there...
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