CUCM 10.5 outbound call from SIP Trunk to CUBE Early Offer Codec advertise issue

[Jasperliang]

HI

I discovered strange behavior of my CCUM.

Call Flow:

 CUCM -> CUBE (Manage by SP)-> ITPT.

I configured SIP EO on trunk.On the SIP trunk profile I am using Best Effort Early Offer, the Media Termination Point Required box is unchecked.

I also configure regions, location and Codec preference lists. 

Somehow the SDP message in early offer that sent from CUCM only advertise G729. From my understand is the CUCM should advise G722, G711 alaw and ulaw then G729. Is that correct?

Content-Type: application/sdp
Content-Length: 264
v=0
o=CiscoSystemsCCM-SIP 14279270 1 IN IP4 10.224.139.101
s=SIP Call
c=IN IP4 10.224.139.101
b=TIAS:8000
b=AS:8
t=0 0
m=audio 27612 RTP/AVP 18 101
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

What would be the reason behind this?

Is the checkbox in SIP profile Allow multiple codecs in answer SDP would help?

Thanks

Jasper

[Aseem Anand]

Hi Jasper,

Can you let me know from where the call is coming into CUCM from? Is it from an IP phone (if yes then its SIP or Skinny) or is the call coming in from another trunk? 

Aseem

[jasper.liang]

HI Aseem

The Call come from another SIP trunk Which is from CME to CUCM.

The completed Call flow is below:

SCCP phone --->CME--->SIP trunk----->CUCM --->SIP Trunk---> CUBE (Manage by SP)--> ITPT.

----------------------------------------------------->G729

CME to CUCM is running G729 between site.  MRGL is configured and applied under both SIP trunks.

[Aseem Anand]

Hi,

Since the call on the incoming call leg is G729 therefore CUCM is sending only G729. If you advertize all the codecs in the invite from CME using voice class codec defined on the outgoing dial-peer, then you will see the CUCM advertising all the codecs. 

Other way to make it work would be by inserting an MTP in the call.

Aseem

[jasper.liang]

Thanks for the advice, Aseem.

I have try this before. And CUCM still only offer one codec(G729) to TIPT. I just added the voice class codec back in. Below is the sdl log from CUCM.

Invite from CME

Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 306
v=0
o=CiscoSystemsSIP-GW-UserAgent 2 6616 IN IP4 10.176.15.254
s=SIP Call
c=IN IP4 10.176.15.254
t=0 0
m=audio 20250 RTP/AVP 18 8 0 101
c=IN IP4 10.176.15.254
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Invite sent to TISP

Content-Type: application/sdp
Content-Length: 252
v=0
o=CiscoSystemsCCM-SIP 25742953 1 IN IP4 10.224.139.102
s=SIP Call
c=IN IP4 10.224.139.101
b=TIAS:8000
b=AS:8
t=0 0
m=audio 29270 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Thanks

Jasper

[Aseem Anand]

Hi Jasper,

You need to check on the CME as to why it is sending the G729 codec only. Check the voice class codec applied on the incoming and outgoing dial-peers on CME.

Run debug voice ccapi inout and check the inbound and outbound dial-peer matched on the CME.

Aseem

[Jasperliang]

HI Aseem

The CME is sending 3 codecs through. I may not explained that clear enough .I have highlighted again.CME  is sending 3 codecs to CUCM G729 G711alaw and G711ulaw. On 183 session CUCM pick only G729.

Invite from CME

Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 306
v=0
o=CiscoSystemsSIP-GW-UserAgent 2 6616 IN IP4 10.176.15.254
s=SIP Call
c=IN IP4 10.176.15.254
t=0 0
m=audio 20250 RTP/AVP 18 8 0 101
c=IN IP4 10.176.15.254
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Correct me if i am wrong. From my understand is CUCM is just the middle man (Provide call controlling and transcoder) to connect the CME and TISP. The CUCM only pick G729 because only one codec is allow it during the call.

I have setup G729  running between CUCM and CME then G711 running between CUCM and TISP. Is the Xcoder miss configured on CUCM will causing the issue?But I have got all the MRGL applied under device pool and SIP trunk.

Thanks 

Jasper

[Evgeny Izetov]

What is the region configuration between the CME trunk and the CUBE trunk? If they are in different regions, then the default is 8kbps, so CUCM will filter out any codec above that.

[Jasperliang]

Hi Evgeny

Thank you! The originate region on CME-CUCM Device pool is running G729 to everywhere,except the site must running G711.

I have changed the Device Pool region on the CME-CUCM trunk to use the same region as CUCM-TISP. That resolved the single codec issue between CUCM and TISP. However, the CME to CUCM is running g711ulaw now which isn't what I want to achieve. Because we only have limited bandwidth between CME and CUCM. 

Is that possible that I can get G729 running between CME-CUCM but the same time the early offer from CUCM to TISP can see all three codecs?

below is the CME voice class codec. 

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

As you can see the ulaw is on preference 3.I understand the 183 session that send back from CUCM decides to choose which codec to talk with CME.Could I change the codec preference list to achieve it(use G729)? Or is that any other way to do that? Thanks again.

Jasper

[Aseem Anand]

Hi Jasper,

If you want to make sure that from CUCM to ITSP the codec being advertised should include G711 as well then one way to achieve this would be by invoking an MTP on trunk with "MTP Preferred Originating CodecRequired Field" set to G711ulaw but then it would consume DSP resources. You would need an IOS based MTP/transcoder and make sure you define both G711 and g729 codec under the dspfarm.

Aseem

[jasper.liang]

Hi Aseem

Thank.  But we want to avoid to use the MTP on the trunk. Is that a way to do that?Thanks again

Regards

Jasper

[Aseem Anand]

Hi,

As per your requirement MTP is the only way to achieve this.

Aseem

(Please rate if useful)

[sipt]

Hello Jasper,

I would also suggest to choose.

Early Offer support for voice and video calls Mandatory (insert MTP if needed) — A SIP
Profile option, where Unified CM inserts a media termination point (MTP) if the media
characteristics of the calling device cannot be determined (for example, for an inbound Delayed
Offer call forwarded over an Early Offer SIP trunk).
• Early Offer support for voice and video calls Best Effort (no MTP inserted) — A SIP Profile
option, where an Early Offer is sent only if the media characteristics of the calling device can be
determined. If the media characteristics cannot be determined, a Delayed Offer is sent.

give a try and see .

Thanks

Sinto

[Evgeny Izetov]

Could the region of the device pool of your SIP trunk be different than the region of the phone? If so, then the default inter-region max bit-rate setting is 8 kbps.

[jasper.liang]

Hi Evgeny

Thanks for reply. The call is originally from CME to CUCM which is running G729 between site.