H323 SIP Media negiciation issue

[Sindibade 78]

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

[Mohammed al Baqari · Accepted solution]

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

[Sindibade 78]

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

[Mohammed al Baqari]

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

[Sindibade 78]

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.

[Mohammed al Baqari]

Hi, you won't use CME. These commands used to register transcoder with cube.

They are supported on your IOS

[Sindibade 78]

No, Just i have see sccp and telephony service. OK i will try to add this params and check.

Thanks for help.

[Sindibade 78]

why you have specified 10.16.41.129 IP. this is used for H323 Loopback interface registring the CUBE into Gatekeeper. 

[Sindibade 78]

Mohamed,

DSP farm need presence of PVDM module on the router?

[Mohammed al Baqari]

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

[Sindibade 78]

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.