01-03-2016 11:05 AM - edited 03-17-2019 05:23 AM
Hello,
We expect issue on VoIP architecture connecting H323 and SIP ip telephony systems through a Cisco Voice Gateway acting as CUBE.
The Cisco Voice Gateway is registred throught it's loopback interface to H323 Gatekeeper and use it to connect SIP IP telephony system.
The issue is related to media codec negociation only when make outgoing calls from SIP to H323.
On the Cisco voice Gateway RTP connection the two peers negociate a different codec bellow:
#sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 5016 5017 17828 2326 10.16.40.2 10.1.6.248
2 5017 5016 16804 0 10.16.41.129 10.16.41.129
Found 2 active RTP connections
show call active media compact
<callID> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 2
5016 ANS T80 g729br8 VOIP P450787 10.1.6.248:2326
5017 ORG T80 g729r8 pre- VOIP P115406 0.0.0.0:0
The calls is setup but no audio heared on both side.
On SIP Telephony system we have fixed only the g729 codec on trunk and IP Phone (10.1.6.248). Wireshark trace only Audio G729 from ip phone (10.1.6.248) to it's SIP Gateway (Cisco Gateway) 10.16.41.129.
No routing issue is expected (each endpoint is reacheable on the network)
From H323 to SIP RTP packet are exchanged correctly and two audio is OK:
sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 5014 5015 17180 32000 10.16.41.129 10.44.29.2
2 5015 5014 16752 2324 10.16.40.2 10.1.6.248
Found 2 active RTP connections
It's not a issue of early media negociation?
Please any recomendation for this issue for a similare experience.
Best regards
Solved! Go to Solution.
01-03-2016 10:17 PM
Hi,
I think the problem is with your codecs. One call leg is using Annex-B and the the one is using No-Annex B. Unless CUBE is configured with transcoder, audio won't flow.
Please configure xcoder in CUBE and register it with telephony-service or CUBE LTI. This will cause tube to invoke xcoder
01-03-2016 10:17 PM
Hi,
I think the problem is with your codecs. One call leg is using Annex-B and the the one is using No-Annex B. Unless CUBE is configured with transcoder, audio won't flow.
Please configure xcoder in CUBE and register it with telephony-service or CUBE LTI. This will cause tube to invoke xcoder
01-04-2016 01:58 AM
HI Mohamed,
Thanks for reply. annexe B was forced in CUB/SIP configuration. so without it we can't get two ways calls for incomming and outgoing H323/SIP calls.
Her is the configuration already implemented in the CUBE:
voice service voip
srtp fallback
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
h323
session transport udp
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 1200 min 300
g729 annexb-all
no call service stop
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
!
interface Loopback1
ip address 10.16.41.129 255.255.255.252
no ip redirects
no ip unreachables
h323-gateway voip interface
h323-gateway voip id Makhazine ipaddr 10.16.7.200 1718
h323-gateway voip h323-id Makhazine@data.net
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.16.41.129
dial-peer voice 200 voip
destination-pattern ......
progress_ind setup enable 3
progress_ind progress enable 8
redirect ip2ip
rtp payload-type g726r16 dynamic
rtp payload-type g726r24 dynamic
voice-class codec 1
session target ras
!
dial-peer voice 108 voip
preference 1
destination-pattern 4507..
redirect ip2ip
rtp payload-type g726r16 dynamic
rtp payload-type g726r24 dynamic
voice-class codec 1
session protocol sipv2
session target ipv4:10.16.41.10
session transport udp
dtmf-relay sip-notify
!
!
gateway
timer receive-rtp 1200
!
sip-ua
!
Thanks
01-04-2016 02:32 AM
In this case, change the order of your voice class and make Annex-B preference one.
Option 2 (below config should fix your problem if option one didn't work):
dspfarm profile 1 transcoder
codec g729r8
codec g729br8
max sess 5
associate application sccp
no shut
!
sccp local lo1
sccp ccm 10.16.41.129 iden 1 verio 7
sccp
!
sccp ccm group 1
assoc ccm 1 pri 1
assoc pro 1 register cube-xcoder
bind source lo1
!
telephony service
max-e 5
max-d 5
ip source address 10.16.41.129
sdspfar uni 1
sdspfar transcode sess 5
adspfar tag 1 cube-xcoder
01-04-2016 03:22 AM
Mohamed,
g729 annexb-all in SIP voip service is not forcing the negociation of g729br8?
What provide the configuration sent? sccp is not used in our case. our Gateway is just a CUBE. no CME conguration into it.
I'm using Cisco 3845 with Advanced Enterprise IOS Version 12.4(24)T2. is this commands are supported on?
Thanks.
01-04-2016 04:24 AM
Hi, you won't use CME. These commands used to register transcoder with cube.
They are supported on your IOS
01-04-2016 04:39 AM
No, Just i have see sccp and telephony service. OK i will try to add this params and check.
Thanks for help.
01-04-2016 08:07 AM
why you have specified 10.16.41.129 IP. this is used for H323 Loopback interface registring the CUBE into Gatekeeper.
01-04-2016 02:21 PM
Mohamed,
DSP farm need presence of PVDM module on the router?
01-04-2016 08:52 PM
Hi,
It is preferred to use loopback to avoid physical interface flapping. Also, you need PVDM for xcoder.
Did you try to change the order of codecs in voice class or even keep G729br8 only? This might resolve the problem if you don't have PVDM
01-06-2016 01:14 PM
Just by changing the codec order. we can get audio but we need to process call hold and resume first on SIP phone to get audio in two ways. My be still have issue to negociate media wih H323.
sh call active media compact ##### the same output while receiving ansering the call and after hold/resume.
<callID> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 2
3017 ANS T123 g729br8 VOIP P115406 10.44.29.2:32004
3018 ORG T123 g729br8 VOIP P450787 10.1.6.248:10020
I attach the debug log call between SIP and H323 for H224 and H225 and SIP Msg.
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