Enterprise VoIP peering is attractive. The incompatibility of IP trunking for different vendor’s solutions begs for solutions beyond the SIPconnect recommendations for SIP trunking to the PSTN. The enterprise peering discussed in the previous blog, "Enterprise VoIP Peering and SIP Trunking, Benefits and Barriers", is further expanded in this blog. None of the solutions discussed in that blog are an out-of-the-box answer for dealing with SIP trunking between two different vendors' call managers.
Although there will many different vendors for interconnection, I will use a discussion of an Avaya-to-Cisco interconnection as an example. There will be some customization or incomplete support in all of the Cisco-to-Avaya solutions. Some of the following issues may also be relevant to SIP trunking between two IPT systems from the same vendor. Here are some of the issues faced by the enterprise:
1. Any solution will have to be re-certified every time either Cisco or Avaya offers a new software release for their respective call managers.
2. There may be media conversion problems with some of the Cisco-to-Avaya IP phone transmission. Some testing will need to be performed. Cisco or Avaya may already have done some of the necessary testing. Whoever does the media testing should be required to officially document the test results.
3. Some telephony features do not match up when reviewing the Cisco and Avaya products. Cisco has one or two call transfer features while Avaya has five transfer features. This mismatch may not be a problem, but the enterprise should test these and other features that could cross the boundaries between the Cisco and Avaya call managers. Setting up and controlling multiparty conference calls is another example.
4. It appears none of the possible solutions has been traffic tested to a saturation level that could possibly occur on a large enterprise network. The solution provider should be tasked with simulating the capacity for the number of simultaneous calls supported across the solution.
5. There will be some development and considerable configuration work no matter what solution is selected.
6. Finding any organization that has a working solution, vendor or enterprise, that has successfully connected Cisco to Avaya call manager over SIP trunks has been difficult. There have been H.323 connections working between their call managers.
7. The users of Cisco phones will not be able to invoke Avaya telephony features and visa versa. The user telephony feature sets will be limited by the call manager (Cisco, Avaya) that the IP phone is registered with.
8. One or more persons will need to become competent and familiar with SIP and the solution, otherwise the enterprise will be tied to the fees and competence of the solution provider.
9. Since a SIP-to-SIP signaling appliance will be strategic to the enterprise’s network, the solution should be highly available and resilient. Some legacy trunking should continue to be in place in case of the SIP trunk solution failure.
Assuming that an enterprise proceeds with a SIP interconnection, extensive testing of the interface between the two IPT systems is necessary whether the IPT systems are from the same or different vendors. The following list of potential tests is not in any order of importance. Some of the tests may be inappropriate for your specific case. The list demonstrates that SIP interoperability testing goes far beyond just testing the SIP protocol operation.
These tests and recommendations are suggested so that all of the features and functions of a network of different vendor's IP PBXs will deliver the functionality that will be required by the enterprise voice users.
1. Call origination and termination
2. Protocol compatibility above UDP
3. Authenticating users across firewalls
4. Demonstrate the operation of the features defined in SIPPING 19
5. Demonstrate the operation of the features defined as extensions by each call manager vendor. These are most likely proprietary features.
6. All initiation and response message validation
7.
Verify the operation of any protocol extensions 8. Demonstrate DID and DOD functionality across the SIP trunk
9. Demonstrate Unified Messaging pass-through
10. Verify that caller ID is conveyed and accurate
11. Verify that a single dial plan encompasses both Cisco and Avaya call managers
12. Call Admission Control compatibility, bandwidth management and conflict resolution
13. Validate the maximum number of calls simultaneously supported by the product installed
14. Verify the maximum Busy Hour Call Completions (BHCC)
15. Determine the maximum number of simultaneous calls possible with the solution
16. Determine the maximum number of SIP endpoints per system
17. Demonstrate how a lost SIP trunk is alerted/alarmed
18. Review the troubleshooting capabilities for the interconnecting SIP trunk
19. Successfully work with NATs and PATs for SIP trunking passing through of firewalls
20. Demonstrate SIP trunking troubleshooting techniques
21. Verify methods for formulating protocol messages for existing options
22. Demonstrate trunk security such as operating over TLS between call managers
23. Verify support of signaling and speech encryption
24. Demonstrate the security authentication and authorization between servers
25. Demonstrate conference bridging set up and termination involving the two different vendors' call managers
26. Determine the maximum number of callers allowed on a bridge
27. Demonstrate the correct support for QoS from end-to-end
28. Determine the extent of RTCP and RTCP XR reporting; are all the fields completed and have the identical meaning?
29. Perform illegal protocol functions without crashing the SIP trunk
30. Try to crash the trunk to see what happens in response, alerts and alarms
31. Call survivability when a SIP signaling connection is lost
32. Verify the complete and accurate creation of Call Detail Records (CDR)
33. Look for differences in the two vendors' CDR contents that may limit their utility
34. Demonstrate the methods and techniques for fault and performance troubleshooting
35. Measure the call setup, disconnect, flash and conference bridging execution times for acceptable performance
36. Measure the voice quality performance and collection of RTCP and RTCP XR messages and alternative implementations by each vendor for reporting voice quality performance
37. Demonstrate the support of the minimum requirements and support for:
a. Codec (G.711 and G.729)
b. Packetization intervals
c. Packet rate
d. Call rate
e. Fax transmission
f. Modem transmission
g. Other device support (alarms, TDD...)
h. 911/E911 call routing
i. Echo cancellation
j. Capability negotiation
38. DTMF tone transport for voicemail and IVR operations (there are 5 different versions supported with SIP trunking)
39. Verify the calling and called name/number displays are correct.
A long list of free VoIP and SIP testing software is available at http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software.
