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SIP-to-H323

Hello,

I have the following configuration:

Asterisk<--(SIP with G711 codec)-->Cisco 2811(ipvoice IOS)<--H323 with g729r8 codec)-->VOIP Provider

I am calling from Asterisk to VOIP Provider. Cisco 2811nm act as SIP/H323 gateway and as G711/G729 transcoder. The problem: when i call from asterisk to VOIP Provider the connection is established but i can not hear the other side and they can not hear me.

A part of config:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none

h323

!

dial-peer voice 2 voip

session protocol sipv2

incoming called-number .T

codec g711alaw bytes 80

!

dial-peer voice 7 voip

destination-pattern 8T

translate-outgoing calling 4

session target ipv4:

tech-prefix 70

codec g729r8 bytes 40

!

gateway

timer receive-rtp 1200

How i can solve this problem ?

15 REPLIES
Hall of Fame Super Gold

Re: SIP-to-H323

You need a transcoding configuration for this, search "configuring transcoding". Also needed, an _ivs image (CUBE) and licence.

If you use asterisk as IP PBX try the router instead.

New Member

Re: SIP-to-H323

I am using c2800nm-ipvoice_ivs-mz.124-18 IOS.

I read this link http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html but i do not understand why i need to configure SCCP ?

What you mean when said:

>If you use asterisk as IP PBX try the router instead.

Hall of Fame Super Gold

Re: SIP-to-H323

You need to because SCCP is the protocol used by transcoding, due to cisco architecture you cannot do otherwise.

The router can work as a nice PBX for SIP phones, that is called CME, that's why I suggested you use it.

New Member

Re: SIP-to-H323

I cannot refuse asterisk use :)

Can you explain a part of config from http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html:

sccp local FastEthernet0/0

sccp ccm 10.4.20.24 identifier 1 version 4.0 - what it means, whose is the address ?

sccp ccm 10.4.20.25 identifier 2 version 4.0

sccp ccm 10.4.20.26 identifier 3 version 4.0

sccp ip precedence 3

sccp

!

sccp ccm group 988

associate ccm 1 priority 1 - what it means ?

associate ccm 2 priority 2

associate ccm 3 priority 3

associate profile 10 register CFB123456789966 (register - what it means)

associate profile 6 register MTP123456789988

keepalive retries 5

switchover method immediate

switchback method immediate

switchback interval 15

!

dspfarm profile 6 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

Re: SIP-to-H323

If your call is getting set up, and not being torn down automatically, you have other problems.

If the call automatically disconnects when you answer or when it rings, then it is a codec negotiation problem.

Regarding the SCCP question:

In order to transcode some operation/process needs to have knowledge of the calls and be able to talk to the transcoder to tell it what to do. You basically configure CME on the CUBE, and then register the transcoder to itself.

hth,

nick

New Member

Re: SIP-to-H323

at http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008084fe1f.shtml and in other links about transcoding the Cisco Call Manager connecting to the router (CUBE if i am right). But how I can connect the asterisk to the router to transcode from one codec to another?

Hall of Fame Super Gold

Re: SIP-to-H323

New Member

Re: SIP-to-H323

thank you for link

Whether I should use an asterisk IP address in this line:

sccp ccm 10.10.10.2 identifier 1

?

Hall of Fame Super Gold

Re: SIP-to-H323

No, a local address to register to itself.

New Member

Re: SIP-to-H323

ok, i will try

Thank you

New Member

Re: SIP-to-H323

I try to configure my router according http://www.cisco.com/en/US/docs/ios/12_4t/12_4t15/it_unitr.html#wp1053671

but i have some problems - i can not set maximum sessions value and codec value in dspfarm profile 10 transcode section:

(config-dspfarm-profile)#maximum sessions ?

<0-0> Number of sessions assigned to this profile

(config-dspfarm-profile)#codec ?

% Unrecognized command

-----------

#sh dspfarm

DSPFARM Configuration Information:

Admin State: DOWN, Oper Status: DOWN - Cause code: ADMIN_STATE_DOWN

Transcoding Sessions: 0(Avail: 0), Conferencing Sessions: 0 (Avail: 0)

Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 600

Connection check interval 600 Codec G729 VAD: ENABLED

Here is a part of config:

voice-card 0

no dspfarm

dsp services dspfarm

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none

h323

!

sccp local FastEthernet0/1.89

sccp ccm 193.109.89.245 identifier 1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 10 register ASTERISK

keepalive retries 5

switchback method immediate

!

dspfarm profile 10 transcode

associate application SCCP

shutdown

!

gateway

timer receive-rtp 1200

!

!

gatekeeper

shutdown

!

Re: SIP-to-H323

This is because you do not have enough DSPs.

'show diag | i FRU' should include some PVDM2-X modules.

If not, you will need DSP resources to do transcoding.

-nick

New Member

Re: SIP-to-H323

I think that no DSP in this router, isn't it ?

#show diag

Slot 0:

C2811 Motherboard with 2FE and integrated VPN Port adapter, 2 ports

Port adapter is analyzed

Port adapter insertion time 1d06h ago

EEPROM contents at hardware discovery:

PCB Serial Number : FOC091736XY

Hardware Revision : 3.0

Top Assy. Part Number : 800-21849-02

Board Revision : B0

Deviation Number : 0

Fab Version : 06

RMA Test History : 00

RMA Number : 0-0-0-0

RMA History : 00

Processor type : 87

Hardware date code : 20050429

Chassis Serial Number : FCZ0924709H

Chassis MAC Address : 0013.c4dd.39b8

MAC Address block size : 24

CLEI Code : CNMJ7N0BRA

Product (FRU) Number : CISCO2811

Part Number : 73-7214-10

Version Identifier : V01

EEPROM format version 4

EEPROM contents (hex):

0x00: 04 FF C1 8B 46 4F 43 30 39 31 37 33 36 58 59 40

0x10: 03 E7 41 03 00 C0 46 03 20 00 55 59 02 42 42 30

0x20: 88 00 00 00 00 02 06 03 00 81 00 00 00 00 04 00

0x30: 09 87 83 01 31 F1 FD C2 8B 46 43 5A 30 39 32 34

0x40: 37 30 39 48 C3 06 00 13 C4 DD 39 B8 43 00 18 C6

0x50: 8A 43 4E 4D 4A 37 4E 30 42 52 41 CB 8F 43 49 53

0x60: 43 4F 32 38 31 31 20 20 20 20 20 20 82 49 1C 2E

Re: SIP-to-H323

Correct - there are no DSPs.

You would need to get DSPs - they are PVMD2-X. The X can be 8,16,32,48,or 64.

You may want to use the DSP calculator to determine what is appropriate for your transcoding situation:

http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl

hth,

nick

New Member

Re: SIP-to-H323

ok,

thank you for help and discussion !

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