With Robin Cai
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to and ask questions about how to design and troubleshoot Cisco Unified Border Element (UBE) for the enterprise with Cisco expert Robin Cai. You can ask questions on how UBE works in different call scenarios, including normal calls, hold/resume, call transfer, call forwarding, as well as best practices.
Robin Cai is a senior customer support engineer in the Cisco Technical Assistance Center in Sydney. His current role includes configuring, troubleshooting and designing Cisco Unified Border Element Enterprise and Service Provider, voice gateways, and Cisco Unified Communications Manager. He has in-depth knowledge of Cisco and industry signaling protocols. He has more than 11 years of experience working in the telecommunication industry. Previously at Cisco he was a senior software test engineer at the China Research & Development Centre in Shanghai. Before joining Cisco, he worked for UTStarcom as a field support engineer and for Huawei Technologies as a software development engineer. Cai holds a bachelor's degree in computer science from Nanjing University of Science & Technology. He also holds CCIE certification (#26037) in Voice and Routing & Switching.
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Robin might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video sub-community discussion forum shortly after the event. This event lasts through December 21, 2012. Visit this support forum often to view responses to your questions and the questions of other Cisco Support Community members.
Solved! Go to Solution.
Thanks for your question.
I assume you mean the MTP in CUCM SIP trunk configuration and the INVITE was sent from CUCM to CUBE. In UCCX/UCCE environment, when the call is being on hold or transferred to agent/IVR, UCCX/CUCM will send RE-INVITE to update the media few times; when the call finally reaches an agent, they will send UPDATE with Remote-Party-ID/P-Asserted-Identity header to update the caller on information of connected party.
In your case, if there is no MTP, the media path is between CUBE and agent or media server. When media changes, CUCM has to update it by sending RE-INVITE to CUBE. If MTP is involved, CUCM keeps the media leg between MTP and CUBE from beginning to end; when necessary, it just updates another leg of MTP between MTP and agent. However UPDATE messages which does not include any SDP, usually has to be passed from end to end.
If you could provide more details as well as debug logs for these two scenarios, I can help to have a look and verify that.
Please let me know if you have any further questions.
The current GW has already communication with the cucm as is acting as h323 GW.. So no issues with that.
Second the outgoing calls through sip are fine so the sip trunk is fine also
Sent from Cisco Technical Support iPhone App
Also I find you have outbound proxy configured under voice service sip, however I don't think proxy is needed for CUBE to communicate with CUCM. Please modify dial-peer 889 as below:
dial-peer voice 889 voip
no voice-class sip outbound-proxy
Let's see how it goes.
Hi Juan Carlos,
Thanks for your question.
Usually in this setup MTP is not needed, as both inbound and outbound call should work without MTP in mose cases. However MTP might be used and needed in some cases, for example, if we need CUCM to do Early Offer (SDP being included in first INVITE sent from CUCM); or if we need to MTP to do DTMF conversion between RTP-NTE and inband.
Please let me know if it answers your question.
Thanks and regards,
I am hoping that you can help me with some security questions for SIP trunks. I have configured CUBE on a 2900 ISR to link to an Internet Telephony Service Provider and want to make sure that it is secure.