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Community Member

H323 Gateway to SIP for ITSP

Hi,

Hoping someone can assist.

I'm using CUCM with a H323 Gateway to a Cisco 3825 running ADV IP services 15.1

CUCM -> H323 Gateway -> 3825 -> SIP -> ITSP

When I attempt and outgoing call I get "Mandatory_IE_Missing" error in the ITSP's logs. I've had this prob before with the SIP trunk and was able to solve it by globally setting SIP with early offer forced.

What I've read this should fix it for h323 as well, but I'm not getting any luck. I've done it in CUCM under the gateway config, enabling "MTP Required" and fast start under outbound. I've also tried just doing on the router as I did for SIP.

voice service voip

ip address trusted list

ipv4 releveant IPs

media statistics

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

h323

sip

early-offer forced

interface GigabitEthernet0/0                     Interface facing ITSP

ip address 192.168.2.1 255.255.255.0

interface GigabitEthernet0/1                                    Interface facing CUCM

ip address 192.168.1.1 255.255.255.0

h323-gateway voip interface

h323-gateway voip bind srcaddr 192.168.1.1

dial-peer voice 101 voip                H323 Dial Peer

incoming called-number 0.........

dtmf-relay rtp-nte

sesstion target ipv4:192.168.1.1

no vad

!

dial-peer voice 100 voip                Dial peer for ITSP

destination-pattern 0.........

session protocol sipv2

session target ipv4:ITSP IP

dtmf-relay rtp-nte

no vad

I really appreciate any assistance.

1 ACCEPTED SOLUTION

Accepted Solutions
VIP Super Bronze

Re: H323 Gateway to SIP for ITSP

Jaco,

The logs make for interesting reading and here is what is going on...

Call come in from cucm over h323 and out to ITSP via sip. Your SIP INVITE was a DO (delayed offer). WHen your provider sent their response in the 200OK with SDP...There was a big programming error in their media attributes

+++Here is the 200 OK+++++

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF251CD
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
CSeq: 101 INVITE
Contact:
User-Agent: FREE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 223

v=0
o=FreeSWITCH 1392036009 1392036010 IN IP4 192.168.8.1
s=FreeSWITCH
c=IN IP4 192.168.8.1
t=0 0
m=audio 25534 RTP/AVP 0 101 13
a=rtpmap:0 G729/8000-------------------------This is wrong.
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Now each codec in a SDP is defined what by what is called RTP/AVP (Audio Video Profile Number) and here are the some of the definitions

0=G711ulaw

8=G711alaw

18=G729

9=G722

As you can see your provider has mapped them incorrectly ans has given G729 to "0"..

Now th eimplication of this is that Cisco Gateway has intepreted 0 as G711ulaw..Here is the H245 trace..TCS sent to CUCM.....

+++TCS to CUCM, showing G711ulaw+++++

191642: Feb 10 21:45:43.403: H245 MSC OUTGOING PDU ::=

value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
    {
      sequenceNumber 1
      protocolIdentifier { 0 0 8 245 0 7 }
      multiplexCapability h2250Capability :
      {
       

----

  capabilityTableEntryNumber 34

          capability receiveRTPAudioTelephonyEventCapability :

          {

            dynamicRTPPayloadType 101

            audioTelephoneEvent "0-16"

          }

        },

        {

          capabilityTableEntryNumber 27

          capability receiveUserInputCapability : basicString : NULL

        },

        {

          capabilityTableEntryNumber 3

          capability receiveAudioCapability : g711Ulaw64k : 20

        }

Sent:
ACK sip:mod_sofia@192.168.8.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF263F4
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Date: Mon, 10 Feb 2014 19:45:43 GMT
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 3779 7493 IN IP4 192.168.2.1
s=SIP Call
c=IN IP4 192.168.2.1
t=0 0
m=audio 16540 RTP/AVP 0 13 101
c=IN IP4 192.168.2.1
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

  capabilityTableEntryNumber 34
          capability receiveRTPAudioTelephonyEventCapability :
          {
            dynamicRTPPayloadType 101
            audioTelephoneEvent "0-16"
          }
        },
        {
          capabilityTableEntryNumber 27
          capability receiveUserInputCapability : basicString : NULL
        },
        {
          capabilityTableEntryNumber 3
          capability receiveAudioCapability : g711Ulaw64k : 20
        }

And CUCM responds with G711ulaw...

++++Next CUBE then sends an ACK to your provider with G711ulaw++++

Sent:
ACK sip:mod_sofia@192.168.8.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF263F4
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Date: Mon, 10 Feb 2014 19:45:43 GMT
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 3779 7493 IN IP4 192.168.2.1
s=SIP Call
c=IN IP4 192.168.2.1
t=0 0
m=audio 16540 RTP/AVP 0 13 101
c=IN IP4 192.168.2.1
a=rtpmap:0 PCMU/8000--------------------------CUBE's 0 is mapped to G711ulaw
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This is the problem...

Suggestions...

1. You need to speak to your provider to correct their programming on their Free Switch. The payload Type for G711 is wrong.

2. You could configure early offer with a different codec from G729 and see if that works. I suggest you try G711alaw.

NB: You will need MTP to do fastart in CUCM for EO to work with SIP.

I think you get on the phone right now with your provider...Thats the best option.

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
14 REPLIES
Hall of Fame Super Silver

Re: H323 Gateway to SIP for ITSP

This message does not necessarily mean it's an issue, is the call failing?

Sent from Cisco Technical Support iPhone App

Community Member

H323 Gateway to SIP for ITSP

Hi,

Yes the call fails, sorry  should have added that.

VIP Super Bronze

H323 Gateway to SIP for ITSP

Jaco,

Your call is failing due a codec mismatch...From the logs

++++ CUBE says no matching codec found++++++

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Info/sipSPIDoAudioNegotiation: No matching voice codec found for m-line 1

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Error/sipSPICompareRespMediaInfo: Media Negotiation failed with no 18x Media

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Error/ccsip_api_call_connect_media: Media Info Failure

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:65, category:278

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[140], src[6]

Feb  9 13:04:11.791: //140/009DAEA90A00/SIP/Info/sipSPIhandle200OKInvite: ccsip_api_call_connect_media returned: SIP_UNACCEPTABLE_MEDIA_ERR

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

codec preference 3 g711alaw

Feb  9 

Since you didnt sent the h323 logs we dont know whats happening on that leg of the call..However it is most likely a config issue..

Your ITSP has sent a 200 OK with g729 codec. What do you have configured on the h323 dial-peer? Also what is the region setting between the phone and the h323 gateway in CUCM? I suggest you configure a voice class codec as follows on your h323 dial-peer to cucm

voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

H323 Gateway to SIP for ITSP

Hi,

Ive attached the ccapu and ccsip all, with and without the voice class codec.

The device pool is the same for the phones and gateway, I didnt want to complicate things yet.

The Dial peers vefore your suggestion of the voice class codec were both g729.

I really appreciate your help.

Community Member

H323 Gateway to SIP for ITSP

Just to add, the call still fails, but now takes 30 seconds or so and then times out, not fast busy like it use to.

VIP Super Bronze

H323 Gateway to SIP for ITSP

Jaco

I looked at the call log and it says normal call clearing...

What we really need to look at is the h225.h245 negotiation...However before you enable the logs 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..

2. debug voip ccapi inout

2. debug h225 asn1

3. debug h245 asn1

4. debug cssip mess


(such as Putty)

then do the ff:

terminal length 0
show logging

Please attach your sh run too (seperately from the logs)

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

H323 Gateway to SIP for ITSP

Hi,

Thanks, I've done as you said.

What I actually want is G729 negotiated between the CUCM, the router and the ITSP.

VIP Super Bronze

Re: H323 Gateway to SIP for ITSP

Jaco,

The logs make for interesting reading and here is what is going on...

Call come in from cucm over h323 and out to ITSP via sip. Your SIP INVITE was a DO (delayed offer). WHen your provider sent their response in the 200OK with SDP...There was a big programming error in their media attributes

+++Here is the 200 OK+++++

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF251CD
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
CSeq: 101 INVITE
Contact:
User-Agent: FREE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 223

v=0
o=FreeSWITCH 1392036009 1392036010 IN IP4 192.168.8.1
s=FreeSWITCH
c=IN IP4 192.168.8.1
t=0 0
m=audio 25534 RTP/AVP 0 101 13
a=rtpmap:0 G729/8000-------------------------This is wrong.
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Now each codec in a SDP is defined what by what is called RTP/AVP (Audio Video Profile Number) and here are the some of the definitions

0=G711ulaw

8=G711alaw

18=G729

9=G722

As you can see your provider has mapped them incorrectly ans has given G729 to "0"..

Now th eimplication of this is that Cisco Gateway has intepreted 0 as G711ulaw..Here is the H245 trace..TCS sent to CUCM.....

+++TCS to CUCM, showing G711ulaw+++++

191642: Feb 10 21:45:43.403: H245 MSC OUTGOING PDU ::=

value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
    {
      sequenceNumber 1
      protocolIdentifier { 0 0 8 245 0 7 }
      multiplexCapability h2250Capability :
      {
       

----

  capabilityTableEntryNumber 34

          capability receiveRTPAudioTelephonyEventCapability :

          {

            dynamicRTPPayloadType 101

            audioTelephoneEvent "0-16"

          }

        },

        {

          capabilityTableEntryNumber 27

          capability receiveUserInputCapability : basicString : NULL

        },

        {

          capabilityTableEntryNumber 3

          capability receiveAudioCapability : g711Ulaw64k : 20

        }

Sent:
ACK sip:mod_sofia@192.168.8.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF263F4
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Date: Mon, 10 Feb 2014 19:45:43 GMT
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 3779 7493 IN IP4 192.168.2.1
s=SIP Call
c=IN IP4 192.168.2.1
t=0 0
m=audio 16540 RTP/AVP 0 13 101
c=IN IP4 192.168.2.1
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

  capabilityTableEntryNumber 34
          capability receiveRTPAudioTelephonyEventCapability :
          {
            dynamicRTPPayloadType 101
            audioTelephoneEvent "0-16"
          }
        },
        {
          capabilityTableEntryNumber 27
          capability receiveUserInputCapability : basicString : NULL
        },
        {
          capabilityTableEntryNumber 3
          capability receiveAudioCapability : g711Ulaw64k : 20
        }

And CUCM responds with G711ulaw...

++++Next CUBE then sends an ACK to your provider with G711ulaw++++

Sent:
ACK sip:mod_sofia@192.168.8.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bKF263F4
From: <0861111111>;tag=6F7D6D4-505
To: <081234567>;tag=1j7UNrHc69Nyp
Date: Mon, 10 Feb 2014 19:45:43 GMT
Call-ID: C3120CDE-91C211E3-819EE519-28587FAF@192.168.2.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 3779 7493 IN IP4 192.168.2.1
s=SIP Call
c=IN IP4 192.168.2.1
t=0 0
m=audio 16540 RTP/AVP 0 13 101
c=IN IP4 192.168.2.1
a=rtpmap:0 PCMU/8000--------------------------CUBE's 0 is mapped to G711ulaw
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This is the problem...

Suggestions...

1. You need to speak to your provider to correct their programming on their Free Switch. The payload Type for G711 is wrong.

2. You could configure early offer with a different codec from G729 and see if that works. I suggest you try G711alaw.

NB: You will need MTP to do fastart in CUCM for EO to work with SIP.

I think you get on the phone right now with your provider...Thats the best option.

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
VIP Super Bronze

Re: H323 Gateway to SIP for ITSP

Jaco,

Have you had a chance to speak to your ITSP? Any update on this

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Community Member

H323 Gateway to SIP for ITSP

Hi

Thank you so much, I got very involved in this and spent a lot of time with them to try and debug.

I learnt a lot thanks to you.

I've gotten to the point now where if I set the voice class codec on the dial peer facing the ITSP, the call goes through 100% with my preferred codec G729.

If I just put G729 codec on my Dial Peer the call fails. This is what they send me now since the made the changes.

v=0

o=FreeSWITCH 1392186288 1392186289 IN IP4 192.168.8.1

s=FreeSWITCH

c=IN IP4 192.168.8.1

t=0 0

m=audio 27960 RTP/AVP 18 0 8 101 13

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

I'm alright with this workaround.

If I do a CUCM -> SIP Trunk -> CUBE -> SIP TRUNK -> ITSP with "early offered forced" set globally on the CUBE, the call would go through as you said it would, even before the changes the ITSP made.

Somehow with the H323 trunk this did not work.

With Early offer on the SIP trunk I also had some strange problems with early ringing.

VIP Super Bronze

H323 Gateway to SIP for ITSP

Jaco,

Thats excellent news. Glad I helped. Can you confirm that the PT (0,18,8) map correctly to the right codec as below in the traces

0=G711ulaw

8=G711alaw

18=G729

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Re: H323 Gateway to SIP for ITSP

Hi Jaco,

Can you check the region setting between the CUCM and the CUBE?

Regards

Kevin

Community Member

H323 Gateway to SIP for ITSP

They couldnt get that to work for some or another reason, so they ended up upgrading to another version of Freeswitch.

Now their 200 ok is sent without that rtpmap altogether.

v=0

o=FreeSWITCH 1392186288 1392186289 IN IP4 192.168.8.1

s=FreeSWITCH

c=IN IP4 192.168.8.1

t=0 0

m=audio 27960 RTP/AVP 18 0 8 101 13

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Which then allows me to use the voice class codec (to send my preference as G729 in reply, I'm guessing with my limited knowledge) and the call to continue.

VIP Super Bronze

H323 Gateway to SIP for ITSP

Yes thats correct..Since G729 is your preffered codec, the gateway will send that to them. I have never seen this type of issue before, so I am really happy we got it fixed!

Good luck. One advice is to use sip to sip not sip to h323. It works better and easier to troubleshoot

Please rate all useful posts

"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
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