07-21-2005 06:22 AM - edited 03-15-2019 03:35 AM
Hi all
I have problems with one-way RTP stream using ATA186 although other IP phones work well. I have network as shown in attached picture -ATA e164 number is 226006675. ATA is loaded with H.323 software (v3.1.2atah323) and has IP adress 10.99.99.4. Default GW is 10.99.99.1 (C2651XM). When calling from ATA to PSTN signaling started, codec g711alaw is designated, channel is opened but audio is only one way (from ATA to PSTN is OK, from PSTN to ATA RTP doesnt goes through). I tried to troubleshhot this problem with ethereal and prserv (both files are attached). In prserv messages "OnReceivedAckPDU: sessionID missing" and "Unknown Q931: 110" are shown. BAsic configuration of ATA is also attached (connect mode, audio mode..).NAT is not problem - no NAT configured. Under voice service voip is fax protocol pass-through g711alaw configured. To avoid problems with firewall I moved ATA before Firewall. I tried almost everythink. Could anyone help me please?? Thank very much.
Regards Miroslav
ON 2951XM are these dial-peers for ATA.
IN: dial-peer voice 989 voip
incoming called-number .T
codec g711alaw
OUT: dial-peer voice 266 voip
huntstop
destination-pattern 226006675
session target ipv4:10.99.99.4
dtmf-relay cisco-rtp
codec g711alaw
fax rate disable
fax protocol pass-through
g711alaw
no vad
=== OUTPUT FROM PRSERV.EXE =============
logging started Thu Jul 21 08:49:14 2005
226006675 active @0xa636304 (GK @0x25642025)
[0]StartTone 0
[0]DTMF 7 , insum:523126
[0]StopTone
[0]DTMF 3 , insum:494590
[0]DTMF 6 , insum:514513
[0]DTMF 6 , insum:498982
[0]DTMF 9 , insum:512845
[0]DTMF 9 , insum:503774
[0]DTMF 0 , insum:514411
[0]DTMF 0 , insum:501916
[0]DTMF 7 , insum:521086
11:00;0,0,0,0,
Calling 736699007
SCC->(0 0) <cmd 16>
CLIP
SCC->(0 0) <cmd 2>
<0 0> dial<736699007>
block queue <- (18 1516504 0)
Connect to <0xa636301 1720>..
>>>>>>>> TX CALLER ID : 0x1 0x80 11
Q931<-0:Setup:CRV 32762
Q931->0:Proceeding
[0]Received pi=8 in q931
Q931->0:Alerting
Connect H245...
block queue <- (19 1516504 557050)
NuConnectDispatcher: 0x7ffa
H245 TCP conn a636301 57538
CESE/MSDSE start:<0 0 0 0>
capSize = 3
H245->0:Cese
[0]: OOB DTMF
RmtInputCap <15 5>
RmtInputCap <15 4>
RmtInputCap <15 1>
RmtAudioCap <4 1>
MD/FRM 1 20
Capability set accepted
H245->0:MSD: <rn tt> = <0xbeb 60>
[0]Slave Tx:1 TxRemote:1
H245->0:CeseAck
H245->0:MsdAck
h323.c 2118: cstate : 4
->H245<0> OLC
H245<-0:LcseOpen
TxAud = G711 (1) 20 fpp
G.711 Silence Suppression on
H245->0:LcseOpen
H245->0:OLC mode 1
remote OpenLogicalReq G711/G729(1) : 20 fpp
OpenRtpRxPort(0,0x0,16384):2
RTP Rx Init: 0, 0
RTP->0:<0xa636304 16384>
H245->0:LcseOpenAck
OnReceivedAckPDU: sessionID missing
RTP<-0:<0xa636301 16596>
[0]Enable encoder 8
[0]: EC 1
RTP TX[0]:SSRC_ID = d3d8f120
07-21-2005 09:16 PM
Miroslav,
Were you able to fix this problem ? I see you have a VOIP dial-peer to ATA (since its using H323) and another VOIP dial peer to your service provider. Normally all gateways are supposed to only connect a POTS to VOIP dial-peer by default. IOS is not enabled by default to connect two VOIP dial peers. So what you need to do is enable the following command in the router to connect two VOIP legs which is typically supposed to be done on a IPGWGW feature set IOS.
voice service voip
allow-connections h323 to h323
redirect ip2ip
HTH
10-25-2005 01:12 PM
Hi,
I tried these commands but with no success. RTP still one way. Have you any other advise, please?
Many thanks for any reply.
11-05-2005 06:19 AM
Really lost
Could anyone help me???
THX
MV
11-05-2005 12:12 PM
Miroslav,
Is this is a Callmanager or Callmanager express environment ?
11-05-2005 12:44 PM
Also, can you paste your configs on voice gateway ?
11-05-2005 01:19 PM
Here you are config of my voice GW where CCME is running (212.90.248.246). Unfortunately I am not able to post config of second side GW, where all my call are sended (213.235.87.71), because I have no access to this GW. But I think problem is on my side :o(
Thank very much for your ideas
11-05-2005 10:16 PM
According to your configuration, and confirmed by your sniffer trace, the router is acting in an "IP-IP" gateway role. This means it plays a proxy role in the middle of the call, helping with H.323 call setup and fixing up the IP addressing in RTP packet headers as they pass through the gateway.
We can assume that your VoIP provider is probably sending RTP traffic properly. You say that calls from IP phones work, and we are seeing RTCP reports from your VoIP provider come in that indicate they're sending RTP. For reasons unknown, that RTP traffic isn't making it to your ATA.
I don't know if it'll help (and it might even hurt, so try it after business hours), but you can try adding the following to your config:
voice service voip
media flow-around
Since your ATA is using private addressing, you may have to move your ATA to back behind your firewall or reconfigure your router's NAT ACL. In a media flow-around config, some sort of NAT will be required since the ATA will now be trying to exchange media traffic directly with your provider's VoIP gateway.
If this doesn't work, we need to see new sniffer traces from inside near the ATA and also outside your gateway if possible (the Internet side). We also need the output of:
debug voip ipipgw
debug cch323 all
debug h225 asn1
debug h225 events
debug h225 q931
debug h245 asn1
debug h245 events
And while a call is up:
show call active voice brief
When you are posting configurations in the future, you may wish to remove passwords, crypto keys, SNMP community strings, etc. If you use 'show tech-support' on the router instead of 'show conf', most of this will be done for you.
11-10-2005 01:53 AM
11-10-2005 09:42 AM
Quick observation here, your call is matching this peer coming in:
dial-peer voice 989 voip
translation-profile incoming Dovnitr
incoming called-number .T
codec g711alaw
And this peer going out to the ATA:
dial-peer voice 266 voip
huntstop
destination-pattern 226006675
session target ipv4:10.99.99.4
dtmf-relay cisco-rtp
codec g711alaw
fax rate disable
fax protocol pass-through g711alaw
no vad
You usually want the peers to be doing the same thing, and I have had one way voice issues with VAD on on one leg and not the other - not with ATA specifically, but it makes me suspicious. I seems to be more trouble than it is worth, so I would turn it off on all your peers and test again. It is also on in your ATA, I don't know if that will hurt of if it gets negotiated, to turn it off there set the audiomode to 0x00140014. Then test again. And, maybe someone else will notice something else..
Mary Beth
11-05-2005 01:03 PM
Hi
Thank for reply. It a CallMAnager EXPRESS.
11-05-2005 02:01 PM
Mir,
I am not sure if this is a codec issue. You are using 2 h323 legs (one to ata and one to PSTN gateway- which is a voip gateway). Your ATA is using g711alaw while your PSTN dial-peer has not specified any codec (which means it defaults to g729).
Can you set the ATA side to g729 and see if that helps ?
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