SIP-to-H.323 Connections fail

[mkt.cisco]

Dear all,

I can't make a following call.

SIPGW -> C2821 -> CUCM -> H323GW

H323GW returns "no resource (47)". Codec is G.729r8.

C2821 configuration is like,

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

sip

dial-peer voice 3 voip

description == H323 Side==

destination-pattern 1111....

session target ipv4:1.1.1.1

dtmf-relay h245-alphanumeric

no vad

dial-peer voice 11 voip

description == incoming call ==

incoming called-number .T

no vad

The opposite call is normal.

Any advices will be helpful. Thnaks in advance.

[Adrian Saavedra]

Hi friend,

Can you check codec settings? May be you can do a debug voice ccapi inout in your gateways.

In this link, you can find codec negotiation values (from debug voip ccapi inout).

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094045.shtml#codecnegvalues

Regards,

- adrián.

[Adrian Saavedra]

Hi,

Also, if this is a matter of codecs, may be you can check this:

"SIP Gateway G.729 Codec Type Mismatch

IOS Session Initiation Protocol (SIP) gateways are used to treat G.729 codec types G.729r8 and G.729br8 as interoperable, but according to RFC 3555 leavingcisco.com this is not true. IOS SIP gateways compliant to the RFC 3555 specification treat G.729r8 and G.729br8 as different codecs. This can cause codec mismatch problems if configured differently on the end points. This can happen with Cisco SIP end points such as the Cisco ATA 186/188, Linksys devices and SIP phones along with some third party SIP end points.

Solution

In IOS SIP gateways complaint to RFC 3555, you need to specify the exact G.729 type of codec in the configuration. Another solution is to downgrade the IOS to a version which is not RFC 3555 complaint. Refer to Enhanced Codec Support for SIP Using Dynamic Payloads for more information on G.729 codecs on SIP gateways."

Regards,

- adrián.

[mkt.cisco]

Thank you for the reply.

I use surely codec as G.729r8 at the SIP endpoints.

I confirmed it by debug command. The current status is,

SIP sends INVITE with FS, however CUCM sends SETUP to H.323GW without FS.

Our SIP-GW doesn't send any TCS afterwards. Finally H.323GW(Cisco) returns "no resource (47)"

Should SIP-GW send TCS after sending INVITE with FS?

or CM should support FS by its setting?

P.S. Sorry for posting to wrong topics.

[rzanett]

Let me see if I am understanding this correctly:

You have a SIP trunk into the 2821. You then point to CUCM. Is this a SIP or H323 trunk to CUCM? Or are you then sending it to as an H323 to the H323 Gateway?

SIP --> 2821 --> H323 --> CUCM --> H323GW

Or

SIP --> 2821 -->SIP --> CUCM --> H323GW

It appears the first way. Let me know as I have all of the following working:

SIP Trunk --> IP-to-IP GW --> H323 to CUCM

SIP Trunk --> IP-to-IP GW --> H323 to Gatekeeper

SIP Trunk --> IP-to-IP GW --> SIP to CUCM

I had to implement transcoding on the router, as my SIP inbound to the gateway was G729 and I wanted to get it to G711.

Have a great day

[rzanett]

Here are some code snippets:

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

sip

bind control source-interface Loopback10

bind media source-interface Loopback10

Bind your SIP to a loopback.

Bind your H323 to loopback:

interface Loopback10

description CUBE Source Loopback

ip address 10.1.1.1 255.255.255.255

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.1.1.1

Note: There are more commands that I have under this interface when registering this as a gateway to Gatekeeper.

The following are dial peers that I have setup to be able to manipulate how and where I send inbound calls to. The phone numbers have been changed but you will get the point. I am able to send calls in as H323 or SIP or to Gatekeeper, etc.

dial-peer voice 1 voip

description incoming-from Verizon

answer-address .

voice-class codec 1

session protocol sipv2

incoming called-number 188844422..

dtmf-relay rtp-nte digit-drop

no vad

!

dial-peer voice 200 voip

description SIP to Regional CUCM

preference 4

destination-pattern 188844422..

session protocol sipv2

session target ipv4:10.31.255.68

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 201 voip

description H323 To Lab Regional CUCM

preference 2

destination-pattern 188844422..

session target ipv4:10.31.255.68

dtmf-relay h245-alphanumeric

codec g711ulaw

!

dial-peer voice 100 voip

description Outbound to Verizon

destination-pattern .T

voice-class codec 1

session protocol sipv2

session target ipv4:172.30.201.17:5185

dtmf-relay rtp-nte digit-drop

no vad

!

dial-peer voice 205 voip

description to Woodbury CUCM

destination-pattern 1888444222.

voice-class codec 1

session target ras

dtmf-relay h245-alphanumeric

!

dial-peer voice 2 voip

description Incoming from CUCM

incoming called-number .

no vad

I also have transcoding resources setup on the router, which uses a subset of CME.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a008092d6b3.shtml

Cheers,

[mkt.cisco]

Thanks for your information. As you noticed, I use H.323 between 2821 and CUCM.

SIP --> 2821 --> H323 --> CUCM --> H323GW

In fact I need not to use Transcoder. Because codec is G.729r8 between SIP endpoints and H323GW.

[mariocard]

can you provide a router configuration example of what you have working

SIP Trunk --> IP-to-IP GW --> H323 to CUCM

SIP Trunk --> IP-to-IP GW --> H323 to Gatekeeper

SIP Trunk --> IP-to-IP GW --> SIP to CUCM