SIP incoming call with G722-64 codec not working

[n.agtual12345]

Hi, Guys.

Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.

Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar)  but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.

Thanks

[Ayodeji Okanlawon]

Hi,

The disconnect cause code is "100" which  means invalid information element. We need to see more logs to know what IE the cucm doesnt seem to like..so you need to do the ff:

conf t:

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then. enable debugs

debug ccsip meesages

debug h225 asn1

debug h245 asn1

debug voip ccapi inout

Enable session capture to txt file in terminal program.> (such as Putty)

then do the ff:

terminal length 0
show logging

Attach the logs.

Second thing I noticed is that no inbound dial-peer is matched for your call..

 Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0

The inbound dial-peer matched is 0 (default)..So you need to add the ff:

dial-peer voice 1006 voip
 description inbound from SIP Carrier
  session protocol sipv2
  incoming called-number 086151410.
 voice-class codec 1  
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 no vad

[n.agtual12345]

Hi,

Thanks for the reply.

Will get back on the debugs. Need to get the ITSP to make the incoming call.

For the dial-peer for incoming call, have that already. It just that the debug that I sent is when I have not put it in. Sorry for that.

dial-peer voice 1000 voip
 description Outbound to SIP Carrier
 translation-profile outgoing SIPout
 destination-pattern .T
 session protocol sipv2
 session target ipv4:202.147.134.21
 incoming called-number .T
 voice-class codec 1  
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 no vad

[n.agtual12345]

G'Day,

Attached are the debugs.

1. logs_040114_01 is incomfing call to 0861514100. G729r8 codec is included on voice class codec. 0861514100 rings and when answered cannot hear anything so i hang up from 0861514100.

2. logs_040114_02 is outgoing call to 1300306907 from 0861514100. G729r8 codec is included on voice class codec. The call is not establishing and see unauthorized on the debugs.

3. logs_040114_03 is outgoing call to 1300306907 from 0861514100. Removed g729r8 on voice class codec and is able to call.

4. logs_040414_04  is incomfing call to 0861514100 (on the second half). Removed g729r8 on voice class codec and is able to call.  0861514100 rings and when answered cannot hear anything so i hang up from 0861514100.

Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar)  but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.

[Ayodeji Okanlawon]

Comment deleted

[n.agtual12345]

Hi,

Thanks for the recommendations. Please see below for my comments. Will be sending the debugs once done.

1. Remove fast start from your h323 gateway. reset the gateway

  - Removed and reset. Codec is g729.

2. configure early offer on your CUBE (it looks like you are laready doing it though)

voice service voip

sip

early-offer forced

- Configured.

3. Your local xcoder on the cube is not configured properly

under telephony-service, your source ip address should be the ip address of the cube not cucm. NB: The sccp config should point to the ip address of your cucm as it is now

- I think IP address 172.27.6.7  is correct. It is the ip address of gi0/0 on the cube.

telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 25
 sdspfarm tag 1 Local_Xcoder
 max-ephones 1
 max-dn 1
 ip source-address 172.27.6.7 port 2000
 max-conferences 8 gain -6
 transfer-system full-consult
!

[n.agtual12345]

Hi,

Unable to do outgoing calls. When fast-start was disabled from  h323 gateway on cucm. So I enable fast-start back.

Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar)  but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.

Regards,

Noel

[Ayodeji Okanlawon]

I have looked at your logs and here are my observations..

1. When you disabled fast start on CUCM, I asked you to enable early offer on your CUBE, however I dont see this in your logs..

This is the INVITE sent to your ITSP, as you can see, this doesnt contain any SDP, that suggest you are doing delayed offer..

Sent:
INVITE sip:1300306907@202.147.134.21:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC862F
From: <sip:0861514100@amcomvoice.ipsystems.com.au>;tag=AE7F464-1B0F
To: <sip:1300306907@202.147.134.21>
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: 9D30000-BA3911E3-824AF7C2-448C8507@172.21.8.134
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE:  1800
Cisco-Guid: 0011462194-3037647155-0083893506-2887478836
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1396510389
Contact: <sip:0861514100@172.21.8.134:5060>
History-Info: <sip:1300306907@202.147.134.21:5060>;index=1,<sip:1300306907@202.147.134.21:5060>;index=2
Expires: 300
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:1300306907@202.147.134.21:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Content-Length: 0

2. If you are doing DO, then your CUBE needs to send an answer to what your ITSP is offering in its ACK..but this is not happening

Here is what I see..Your CUBE sends SDP in its PRACK

Sent:
PRACK sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@202.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC9232B
From: <sip:0861514100@amcomvoice.ipsystems.com.au>;tag=AE7F464-1B0F
To: <sip:1300306907@202.147.134.21>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: 9D30000-BA3911E3-824AF7C2-448C8507@172.21.8.134
CSeq: 103 PRACK
RAck: 323009643 102 INVITE
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@202.147.134.21:5060;transport=udp",response="a9d772d988ec971cdad556fd4a992bd0",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="58621262",qop=auth,algorithm=MD5,nc=00000002
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 293

v=0
o=CiscoSystemsSIP-GW-UserAgent 1079 7198 IN IP4 172.21.8.134
s=SIP Call
c=IN IP4 172.21.8.134
t=0 0
m=audio 17082 RTP/AVP 8 96 100
c=IN IP4 172.21.8.134
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20

###Here is your ACK to the 200 OK from ITSP###
On the ACK...Your CUBE doesnt include any SDP in its ACK, hence your ITSP disconnected the call immediately

ACK sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@202.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKCA1335
From: <sip:0861514100@amcomvoice.ipsystems.com.au>;tag=AE7F464-1B0F
To: <sip:1300306907@202.147.134.21>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: 9D30000-BA3911E3-824AF7C2-448C8507@172.21.8.134
Max-Forwards: 70
CSeq: 102 ACK
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:1300306907@202.147.134.21:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0

017151: Apr  3 07:33:11.089 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:0861514100@172.21.8.134:5060 SIP/2.0
Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bKahsb3f108gbhq8pbn5k1sdj0gkbf0.1
From: <sip:1300306907@202.147.134.21>;tag=SD7qfu599-1874793413-1396510390487
To: <sip:0861514100@amcomvoice.ipsystems.com.au>;tag=AE7F464-1B0F
Call-ID: 9D30000-BA3911E3-824AF7C2-448C8507@172.21.8.134
CSeq: 323009054 BYE
Max-Forwards: 9
Content-Length: 0

I have two suggestions..

1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour

2. Send your full sh run

On your inbound call issue, you need to send me the logs for a call to another of your DDI..

[n.agtual12345]

Hi,

Thanks again for the reply.

1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour.

    - What ios do you recommended?

   - The behavior is weird. All outbound calls are working (mobile, pstn, itsp's polycom). For inbound calls, only the itsp's polycom phones are not working.

2. Send your full sh run

   - Please see attachment.

   - Itsp ask me to change realm to resolve the incoming called number DDI. But when i change the realm  i'm not able to register so they are looking at it.

[n.agtual12345]

Hi,

Also, early-offer force is enabled on the cube and fast-start on h323gw on cucm was disabled when i capture the logs and calls are not working.

So, I enabled back fast-start on cucm. And calls become to work again.  Except for the incoming call from Itsp polycom phone (original issue).

[Ayodeji Okanlawon]

OK.

Before we go ahead..can I suggest that you change the h323 leg to sip. Create a sip trunk in cucm to your cube gateway and configure your inbound dial-peer to cucm and outbound dial-peer to cucm to use sip.

[n.agtual12345]

So, do I have to route the segment facing the ITSP if I change to sip leg?

Cause the reason I used h323 to sip is that the segment that the ITSP  provided (172.21.8.134/29) is used in our network.

[Ayodeji Okanlawon]

What you need to do is remove the bind statement under gig 0/1 and apply it to the dial-peers instead. Read the blog below. It explains how and why you should do this..

Apply the bind to the dial-peer to the ITSP and to the cucm. Ensure the ip address on the local interfcae for the cube (gig0/0) is what you use on the sip trunk on cucm,

https://supportforums.cisco.com/blog/154506

[n.agtual12345]

G'Day,

Have been able to change to SIP trunk in-between CUCM and the gateway.

Also, ITSP made some changes on their end, and already able to make incoming calls, calling other numbers from the range they given to us (aside from the main number).

However, I'm only able to make outgoing calls using the main number.  Please see logs below.

[Manish Prasad]

Can you test by changing authentication username like this...

 authentication username AMM-4324-Trunk password xxxxxxxxxxxxxxxxxxxxxxx realm sipconnect.amcomvoice.ipsystems.com.au

Thanks

Manish