IPIPGW

smvasagam · ‎08-25-2007

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

jposada_us · ‎09-03-2007

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

smvasagam · ‎09-03-2007

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

vmalhi · ‎09-18-2007

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.

paolo bevilacqua · ‎09-18-2007

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!

vmalhi · ‎09-18-2007

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?

paolo bevilacqua · ‎09-18-2007

Point taken and I stand corrected. Thanks!

vmalhi · ‎09-18-2007

No probs! You had me questioning myself there...