02-08-2012 11:18 AM - edited 03-16-2019 09:28 AM
Hi
I am using Cisco 2811 router to converge 2 RTP streams of different codes. I have scenario as below:
SIP PBX-------g729------>CISCO2811-------g711------>CUBE
I am using the universal transcoder, it does kick-in but no rtp gets established. Could some one please advise?
here is the config:
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
fax protocol pass-through g711alaw
h323
h225 h245-address on-connect
sip
bind control source-interface FastEthernet0/1
bind media source-interface FastEthernet0/1
!
!
!
voice class codec 10
!
voice class codec 100
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
!
!
voice-card 0
dsp services dspfarm
!
!
interface FastEthernet0/0
ip address 62.215.x.x 255.255.255.248
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 10.50.1.42 255.255.255.252
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 62.215.231.49
ip route 80.184.x.x 255.255.255.0 10.50.1.41
!
no ip http server
no ip http secure-server
!
ip nat sip-sbc
ip nat inside source list 2 interface FastEthernet0/0 overload
ip nat inside source static 10.50.1.42 62.215.x.x
!
access-list 2 permit any
access-list 100 permit ip any any log
!
!
sccp local FastEthernet0/1
sccp ccm 10.50.1.42 identifier 1 version 3.1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 110 register MTPNEW
!
dspfarm profile 110 transcode universal
codec g729abr8
codec g711alaw
codec g729ar8
codec g729r8
codec g729br8
maximum sessions 12
associate application SCCP
!
!
dial-peer voice 10 voip
description ***Outgoing Calls to SIP Trunk***
destination-pattern .T
modem passthrough nse codec g711alaw
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:80.184.2x.y
session transport udp
dtmf-relay rtp-nte
codec g711alaw
fax rate disable
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 2 voip
translation-profile incoming inbound
voice-class codec 100
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
no vad
!
!
!
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 MTPNEW
max-ephones 1
max-dn 1
ip source-address 10.50.1.42 port 2000
max-conferences 8 gain -6
call-forward pattern .T
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
!
02-08-2012 10:58 PM
So you mean the call drops or the link remains up but no voice flows ?
GP.
02-09-2012 02:27 AM
Call gets established, I can see the transcoder session, rtp connections. But the audio is one way.
02-09-2012 04:45 AM
aah, thanks. One way audio is mainly due to absense of required access route in both directions. Could you please verify that you have the routing in place for both directions, in 2811 ?
GP.
02-11-2012 06:31 AM
Hi,
I got it working. Now I see that despite having enough DSP resources, no more then 16 calls are getting through. Is there any limitation or configurable parameter that would govern the number of calls on the SIP trunk?
-saif
02-11-2012 07:37 PM
Saif,
But it seems the issue was different earlier ( as mentioned in original description ) ? Earlier, the calls were established but the audio was one-way. Was that fixed & above was a new issue ( i.e. more than 16 calls not getting established ) ?
GP.
02-12-2012 09:52 AM
Yes, the earlier issue of one-way voice was resolved. We had some configuration issue on the IP-IP gateway. I think, I should post next issue on another discussion.
saif
02-12-2012 08:23 PM
Thanks Saif. It would really help readers here if you could elaborate on the config issue you had on CUBE & the fix. Was that routing issue for sure ?
GP.
02-12-2012 11:29 PM
Actually the problem was definition of the SIP 'bind source'. In case of IP-IP gateway, this parameter should be left to the routing table to decide.
saif
02-12-2012 11:42 PM
Thanks Saif. +5 to you.
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