IPIP GW to IPIP GW problem

czerewacz · ‎12-01-2010

Dears:

I'm having a problem with next topology:

Incoming calls: Those going to the CUCM Cluster work fine.

Outgoing calls: Not working. When the customer calls to a DNIS the call is established but there is no audio in any way.

The thing is that if I bypass the IPIP ISP and register the IPIP CUSTOMER directly with the ISP GK, Outgoing calls work fine.So I think that i have a problem with the sip-to-sip between IPIPGW's. Here you have part of configs:

IPIP ISP:

voice service voip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
h323
h245 tunnel disable
sip
bind control source-interface Loopback0
bind media source-interface Loopback0

IPIP CUSTOMER:

voice service voip

address-hiding

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

emptycapability

h225 id-passthru

h225connect-passthru

h245 caps moderestricted

h245 passthrutcsnonstd-passthru

sip

g729 annexb-all

Please help me with this issue. If someone need more info just ask!

Thanks and Best Regars

Marcelo

Abu Hadee · ‎12-01-2010

Hi Marcelo

Its hard to tell where is the problem at this point without looking at some further info.

Please do the following

1. Make a call to PSTN and keep the call active

2. On Customer CUBE, do "show voip rtp connection"

3. on ISP CUBE, do "show voip rtp connection"

now check the local and remote RTP ip address? Does it seems all good? What are those IP Addresses?

4. Now, do "show call active voice brief" on both CUBE couple of times.

Check if the tx and rx are incrementin on all call legs.

Also, if you keep the call long enough, does it disconnect it self later some time? Or it remains active as long as you want?

Please update me with the above information, I may help you finding the root cause or have next course of action.

Thank you

- abu

czerewacz · ‎12-01-2010

Thanks for your answer. It is good stuff so asap i will test that and share my comments!!

czerewacz · ‎12-03-2010

Dear abu:

I could not permormt the test that you mention due to customer availability but i have from old test a sh call active voi brief made in the ISP IPIP with the call established:

1E13 : 114 435495050ms.113 +9150 +595880 pid:1 Originate (destination number)
dur 00:09:46 tx:0/0 rx:10179/196810 10 (normal call clearing (16))
IP 200.12.1.2:17460 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

1E13 : 113 435495040ms.114 +9170 +595900 pid:5 Answer (xxxxx)
dur 00:09:46 tx:0/0 rx:0/0 10 (normal call clearing (16))
IP 190.168.1.6:18310 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

This it an outgoing call that as you can see is established but no audio (Also attached the sh call act voi id).

I have one doubt (by your suggested test): Is the IPIP CUSTOMER have a sip to sip to ISP and also a sip to sip to his CCM: which interface will use if no bind address command is involved?

Thanks and best regards

czerewacz · ‎12-08-2010

Dears: I could ran the commands suggested:

CUSTOMER IPIP#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP LocalIP                                             RemoteIP

1   3630       3631       17944    4000   192.168.1.6 (Local Gi0/0)                         192.168.1.58 (local ip phone)
2 3631       3630       16568    16430 10.10.10.242 (Local Gi0/1)                         10.10.10.26   (ISP IPIP Gi 0/0)

ISP IPIP#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP LocalIP                                RemoteIP
1   41         42         16430    16568 10.10.10.26 (Local Gi0/0)                       10.10.10.242 ( CUSTOMER IPIP Gi0/1)
2   42         41         17878    18456 10.10.10.26 (Local Gi0/0)                        200.156.58.12   (ISP SBC)

These are the sh call act vo brie:

sh call acti voi brief in ISP IPIP

2409 : 42 448645370ms.41 +10940 +135990 pid:2 Originate (called number)
dur 00:02:05 tx:0/0 rx:168/3226 10 (normal call clearing (16))
IP 200.156.58.12:18456 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a
2409 : 41 448645360ms.42 +10960 +136010 pid:5 Answer 82500
dur 00:02:05 tx:168/3226 rx:0/0 10 (normal call clearing (16))
IP 10.10.10.242:16568 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

*****************************************************************************************************

sh call acti voi brief in CUSTOMER IPIP

2409 : 3630 594009320ms.1 +10980 pid:200 Answer internal number connected
dur 00:01:06 tx:0/0 rx:0/0
IP 10.10.10.58:4000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

2409 : 3631 594009320ms.2 +10980 pid:200 Originate (called number) connected
dur 00:01:06 tx:0/0 rx:105/2014
IP 10.10.10.26:16430 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

Do you think that i must bind the Gi 0/1 (10.10.10.242) for the sip-to-sip in IPIP GW?

Please any comments on this will be great!!

Also if anybody knows where i can find CUBE configs examples to connect them back to back!

Thanks!!!