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Cisco Unified Communications Manager SIP Trunking Workshop

Learn the basics of CUCM setup, partitions, calling search spaces, number manipulation, and SIP trunking. Explore early offer vs delayed offer and experience a demo call via the workshop SIP server.

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Cisco Unified Communications Manager SIP Trunking Workshop

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  1. Cisco Unified Communications Manager SIP Trunking Bill Efthimiou APAN33 SIP workshop February 2012 Network Operations

  2. Agenda • Basic CUCM setup • Partitions • Calling Search Spaces • Number Manipulation • SIP trunking • Early offer vs Delayed offer • Demo call via workshop SIP server. 2

  3. Basic CUCM Setup – Activate required services 3

  4. Basic CUCM Setup – Cisco Unified CM Auto Registration 4

  5. Basic CUCM Setup – Cisco Unified CM Group • Redundancy • primary vs secondary 5

  6. Basic CUCM Setup – Phone NTP References 6

  7. Basic CUCM Setup – Date/Time Groups 7

  8. Basic CUCM Setup – Locations (Call Admission Control) Use locations to implement call admission control in a centralized call-processing system. Call admission control enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between the locations. 8

  9. Basic CUCM Setup – Device Pool

  10. Basic CUCM Setup – Enable CDR

  11. Basic CUCM Setup – Media Resource Group

  12. Basic CUCM Setup – Media Resource Group List 12

  13. CUCM Partitions The dial plan entries that you may place in a partition include IP phone directory numbers, translation patterns, route patterns, CTI route points, and voicemail ports. If two or more dial plan entries (directory numbers, route patterns, or so forth) overlap, Unified CM selects the entry with the closest match (most specific match) to the dialed number. In cases where two dial plan entries match the dialed pattern equally, Unified CM selects the dial plan entry that appears first in the calling search space of the device making the call. Source: Cisco Unified Communications System 8.x SRND

  14. CUCM Calling Search Spaces A calling search space defines which partitions are accessible to a particular device. Devices that are assigned a certain calling search space can access only the partitions listed in that calling search space. Attempts to dial a DN in a partition outside that calling search space will fail, and the caller will hear a busy signal. If you configure a calling search space both on an IP phone line and on the device (phone) itself, Unified CM concatenates the two calling search spaces and places the line's calling search space in front of the device's calling search space. Source: Cisco Unified Communications System 8.x SRND

  15. CUCM – Partition Order Source: Cisco Unified Communications System 8.x SRND 15

  16. CUCM – Concatenation of Line and Device CSS Source: Cisco Unified Communications System 8.x SRND 16

  17. Number Manipulation – Route Pattern

  18. Number Manipulation – Translation Pattern

  19. Number Manipulation – Route List Detail

  20. SIP Trunking – CUCM 8.5 SIP trunks provide connectivity to other SIP devices such as gateways, SIP proxies, applications, and other Unified CM clusters. Today, SIP is arguably the most commonly chosen protocol when connecting to service providers and Unified Communications applications. Cisco Unified CM 8.5 and later releases provide the following SIP trunk and call routing enhancements: • Can run on all Unified CM nodes • Up to 16 destination IP addresses per trunk • SIP OPTIONS ping keepalives • SIP Early Offer support for voice and video calls (insert MTP if needed) • QSIG over SIP • SIP trunk normalization and transparency • Supports the use of route lists on all Unified CM nodes The SIP trunk features available in the current release of Unified CM make SIP the preferred choice for new and existing trunk connections. Source: Cisco Unified Communications System 8.x SRND

  21. SIP Trunk Configuration – 8.0

  22. SIP Trunk Configuration – 8.0 (continued)

  23. SIP Trunk Security Profile

  24. Delayed offer vs Early offer • Delayed offer = NO SDP in original SIP INVITE • Early offer = SDP included as part of original SIP INVITE • With delayed offer, caller makes the final codec decision. • With early offer, called makes the final codec decision. • Prior to CUCM version 8.5, SIP trunk only supported delayed offer. • AARNet’s experience with delayed offer was, issues when calling MCUs. • MCU’s support many codec’s. Therefore, the responding SIP/SDP body, • is very large and causes massive fragmentation issues. Signaling or • negotiation would often fail.

  25. Delayed offer- trace 25

  26. Early offer- trace 26

  27. CUCM Early Offer enabled Copy “Standard SIP Profile” Standard SIP Profile  Standard SIP Profile- w early offer. 27

  28. CUCM Early Offer enabled <snip> 28

  29. Thank you Questions?

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