cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1162
Views
5
Helpful
3
Replies

CME RTP Streams

AT
Level 1
Level 1

I am running cme 11.5 with 7841 SIP phones.  A wireshark capture shows RTP streams going through the CME router and not directly between phones.  Is this normal?  Should I see RTP chatter between two SIP phones ?

AT

3 Replies 3

sanjaydevan
Level 1
Level 1

I believe this is normal. Media-flow round is not supported rather Media-flow through in CME. 

Regards

Devan

Please rate useful posts!

Thank You.

Manish Gogna
Cisco Employee
Cisco Employee

Hi AT,

You may check the CME admin guide for info on which type of calls support media flow thru and flow around

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmeadm/cmesystm.html

Media Flow Around Support for SIP-SIP Trunk Calls

Cisco Unified CME 8.5 and later versions support the media flow around functionality for SIP to SIP trunk calls on Cisco Unified CME, allowing less consumption of resources on Cisco Unified CME.

The media flow around feature eliminates the need to terminate RTP and re-originate on Cisco Unified CME. This reduces media switching latency and increases the call handling capacity for a Cisco Unified CME SIP trunk.

Media flow around is supported in the following scenarios:

  • Single Number Reach (SNR) Push—If an SNR call on a SIP trunk is pushed over to a mobile user over another SIP trunk, the resulting connection is a SIP-SIP trunk call connection. If both SIP trunks are configured for media flow around, the media is allowed to flow around Cisco Unified CME for the resulting call.

  • Call Forward—If a SIP trunk call is forwarded over another SIP trunk and both the SIP trunks are configured for media flow around, media flows around Cisco Unified CME for the resulting SIP-SIP trunk call. Media flow around is supported for all types of call forwarding, such as call forward night-service, call forward all, call forward busy, and call forward no-answer.

  • Call Transfer—If a SIP trunk call is transferred over another SIP trunk and both SIP trunks are configured for media flow around, media flows around Cisco Unified CME for the resulting SIP-SIP trunk call. Media flow around is supported on both SIP-line-initiated call transfer and SCCP-line-initiated call transfers. It is supported for all types of call transfers, such as blind transfer, consult transfer, and full consult transfer.

Media is forced to flow through on different types of call flows including the SIP to SIP trunk call with asymmetric flow mode configurations or symmetric flow through configuration. In asymmetric flow mode configurations, one SIP leg is configured in the media flow around mode and another SIP leg is configured in the media flow through mode. In such cases, media is forced to flow through Cisco Unified CME.

Media is forced to flow through Cisco Unified CME for the following types of call flows:

  • Any calls involving a SIP endpoint, a SCCP endpoint, PSTN trunks (BRI/PRI/FXO), or FXO circuits.

  • SIP to SIP trunk call with either asymmetric flow mode configurations or symmetric flow through configurations.

  • SIP to SIP trunk call that requires transcoding services on Cisco Unified CME.

  • SIP to SIP trunk calls that require DTMF interworking with RFC2833 on one side, and SIP-Notify on the other side.

  • SNR pullback to SCCP— When an SNR call is pulled back from a mobile phone to the local SCCP SNR extension, the call is connected to the SCCP SNR extension. Media is required to flow through Cisco Unified CME because one of the calls is from a SCCP SNR extension, which is local to Cisco Unified CME.

In Cisco Unified CME 8.5, the media flow around feature is turned on or turned off using the media command in voice service voip, dial-peer voip, and voice class media configuration modes. The configuration specified under voice class media configuration mode takes precedence over the configuration in dial-peer configuration mode. If the media configuration is not specified under voice class media or dial-peer configuration mode, then the global configuration specified under voice service voip takes precedence. For more information, see Enable Media Flow Mode on SIP Trunks.

HTH

Manish

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: