CUCM SIP Invite not offering SRTP / SAVP

Hi,

I am trying to setup encrypted SRTP via SIP Trunk.

I have already configured the following:

1) SIP Security Profile to use "Encrypted TLS" and enable "transmit security status"

2) SIP Trunk to enable "Allow SRTP with TLS"

3) SIP Profile to enable "early call offer" and "send SDP in mid-invite"

However, I noticed that the SIP invite offered by the CUCM is still RTP.(m=audio RTP/AVP)

What else do I need to configure to get the CUCM to offer SRTP (m=audio RTP/SAVP) in the SIP invite?

Below is the SIP invite from the CUCM:

   INVITE sip:3334@192.168.1.103:5081 SIP/2.0

   Via: SIP/2.0/TLS 192.168.1.105:5081;branch=z9hG4bK126c5b3a27d0

   From: <sip:1807@192.168.1.105;x-nearend;x-refci=29768466;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP001E7AC4D85F;x-nearendaddr=1807;x-farendrefc

i=29768467;x-farendclusterid=StandAloneCluster;x-farenddevice=SEP58BC27756FCC;x-farendaddr=1815>;tag=5621~4e09c267-fb7b-9d1f-0d16-fc861f72f307-29768471

   To: <sip:3334@192.168.1.103>

   Date: Wed, 23 Mar 2016 07:31:29 GMT

   Call-ID: 40b7e400-6f214651-1223-7301a8c0@192.168.1.105

   Supported: timer,resource-priority,replaces

   Min-SE:  1800

   User-Agent: Cisco-CUCM10.0

   Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

   CSeq: 101 INVITE

   Expires: 300

   Allow-Events: presence, kpml

   Supported: Geolocation

   Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Unknown; gci= 1-13038, <sip:192.168.1.105:5081>;method="NOTIFY;Event=telephone-event;Duration=500"

   Cisco-Guid: 1085793280-0000065536-0000000019-1929488576

   Session-Expires:  1800

   P-Asserted-Identity: <sip:1807@192.168.1.105>

   Remote-Party-ID: <sip:1807@192.168.1.105>;party=calling;screen=yes;privacy=off

   Contact: <sip:1807@192.168.1.105:5081;transport=tls>;isFocus

   Max-Forwards: 70

   Content-Type: application/sdp

   Content-Length: 275

   v=0

   o=CiscoSystemsCCM-SIP 5621 1 IN IP4 192.168.1.105

   s=SIP Call

   c=IN IP4 192.168.1.105

   b=TIAS:64000

   b=AS:64

   t=0 0

   m=audio 24614 RTP/AVP 0 8 18 101

   a=rtpmap:0 PCMU/8000

   a=rtpmap:8 PCMA/8000

   a=rtpmap:18 G729/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-15

Thanks.

Sincerely,

Merv

Hello mervkwok1,

Please move the post to the support community.

In this case you need assistantance with CUCM operation and configuration.

try this area:

https://supportforums.cisco.com/community/5961/ip-telephony