MTP not always invoked

csg-admin · ‎09-03-2008

Hi,

My setup is as follows:-

CUCM--> SIP TRUNK --> CUBE --> SIP --> Service Provider

The CUCM SIP is configured for early offer with MTP checked.

MY CUBE config is as follows:-

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

no supplementary-service h450.2

no supplementary-service h450.3

no supplementary-service h450.7

supplementary-service media-renegotiate

supplementary-service ringback h225-info

redirect ip2ip

h323

emptycapability

sip

bind control source-interface GigabitEthernet0/0

bind media source-interface GigabitEthernet0/0

early-offer forced

midcall-signaling passthru

g729 annexb-all

sccp ccm group 2

associate ccm 2 priority 1

associate profile 1 register CUBEXCODE

keepalive retries 1

keepalive timeout 10

switchover method immediate

switchback method immediate

dspfarm profile 1 transcode

codec g711ulaw

codec g729ar8

codec g729r8

maximum sessions 24

associate application SCCP

dspfarm profile 4 mtp

codec g711ulaw

maximum sessions software 500

associate application SCCP

!

dspfarm profile 5 mtp

codec g729r8

maximum sessions software 500

associate application SCCP

!

dspfarm profile 6 mtp

codec g729ar8

maximum sessions software 100

associate application SCCP

dial-peer voice 200 voip

description ** outgoing call to CUCM **

destination-pattern 44T

session protocol sipv2

session target ipv4:x.x.x.x

dtmf-relay rtp-nte digit-drop

codec transparent

!

dial-peer voice 300 voip

description ** incoming dial peer **

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte digit-drop

codec g711ulaw

!

dial-peer voice 400 voip

description ** outgoing call to SIP Provider **

destination-pattern 9T

session protocol sipv2

session target ipv4:x.x.x.x

dtmf-relay rtp-nte digit-drop

codec transparent

I can make G711 calls in and out which always invoke an MTP.

All my G729 calls never invoke an MTP but invoke a transcoder instead. I can transfer these calls to G711 phones and vice versa.

My question is, is this normal behaviour?

Thanks,

gogasca · ‎09-03-2008

SIP Early Offer is restricted for G711 only

SIP negotiates media exchange via Session Description Protocol (SDP), where one side offers a set of capabilities to which the other side answers, thus converging on a set of media characteristics. SIP allows the initial offer to be sent either by the caller in the initial INVITE message or, if the caller chooses not to, the called party can send the initial offer in the first reliable response. By default, Unified CM sends the INVITE without an initial offer, and it requires MTP resources to send the offer in the INVITE. Note that this initial offer is limited to the G.711 codec only.

Also note that MTP resources are not required for incoming INVITE messages, whether or not they contain an initial offer.

For the G729 calls, are they incoming from CCM to CME or viceversa?

Thanks

csg-admin · ‎09-04-2008

Thanks for the reply.

The G729 and G711 phones are on the same CUCM 6.1 cluster but in different regions.

Here is a show sccp connections for both a G729 and G711 outgoing call from CUCM:-

sess_id conn_id stype mode codec ripaddr rport sport

33555870 33564521 xcode sendrecv g729 172.30.15.11 29552 17898

33555870 33564519 xcode sendrecv g711u 10.25.0.50 17490 16738

sess_id conn_id stype mode codec ripaddr rport sport

33555871 33564534 mtp sendrecv g711u 172.30.15.20 24664 18662

33555871 33564533 mtp sendrecv g711u 10.25.0.50 16842 19502

AS you can see G711 invokes an MTP, early offer call, whereas G729 doesn't. I can understand the point that a G729 call will need a transcoder to talk G711 but is the transcoder also acting as an MTP and sending the invite with early offer?

Here is my debug ccsip message and below is the first sdp message for a G729 call:

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.25.0.11

s=SIP Call

c=IN IP4 10.25.0.50

t=0 0

m=audio 16682 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

For G729 I would have expected to see both a transcoder and MTP being used to set up the call so I am a bit confused as to what is going on and whether this config is scalable?

Thanks,

Michael Owuor · ‎09-04-2008

Yuko,

What your are seeing is the expected behavior. In CUCM 6.x, SIP trunks only support G711a/ulaw with pre-allocated MTP.

In the CUCM 7.x, SIP Trunk will allow early-offer calls (calls with pre-allocated MTP) to be initiated with low bandwidth codecs such as G.729.

Here's the Release Note of CUCM 7.0 that calls out the enhancement in this version:

* Support for G.729a and G.729b Codecs Over SIP Trunks

G.729a and G.729b are low-bandwidth codecs that can be used for calls that are initiated over SIP trunks. Be aware that this feature is required for endpoints that do not support delayed media calls and do not want to use a higher-bandwidth codec, such as G.711.

Because an MTP needs to be pre-allocated for early-offer calls, you must configure an external MTP or transcoder device to use this feature. The software MTP does not support G.729 over SIP trunks.

Although this feature supports all four G.729 codecs (G.729, G.729a, G.729b, and G.729ab), the system cannot distinguish between G.729 and G.729a or between G.729b and G.729ab. Therefore, Cisco Unified Communications Manager Administration provides only two options for configuring these codecs on SIP trunks: G729/G729a and G729b/G729ab.

The G.729 codec over SIP trunks applies only to outgoing calls, and incoming calls are not affected. Be aware that the system does not support mid-call codec switching from G.729 to any other codec."

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a08sip.html#wp1221209

Hope this helps.

Regards,

Michael.

csg-admin · ‎09-04-2008

Thank you both for your answers.