Thu 10 Dec 2009
Skype for SIP Beta Program Now Open for Public !
Posted by Tariq Ahmad under Cisco Unified Call Manager Express , Session Initial Protocol(SIP)1 Comment
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With the Skype for sip beta program now in open public domain , there are open integration options available for Cisco Routers esp. CUCME as well as CUCM.Both SIP Digest Authentication as well as IP authentication are supported. For those businesses eager to test integration , you can test service by setting up a Business control Panel (BCP) account.You can avail upto 300 simultaneous voice channels if you are running a large business. Also, if you already have peering with a SIP Service Provider (See Tutorials :GNS3:How to Setup CME SIP trunk to VOIP SIP Service Provider ) , you can integrate another CUCME to skype for sip and integrate both solutions as Skype for sip offers support for
G711 in addition to G729 now.
Here are main standards that Skype for SIP currently supports for integration:
- SIP: Session Initiation Protocol (RFC3261)
- RTP: Real Time Protocol for use with G.711 and G.729 codec’s (20/30/40ms)
- DTMF: Dual Tone Multi Frequency (Inband and RFC2833)
- SIP Digest for authentication of SIP profiles
- Authentication on a call by call basis when using SIP registrations
- IP authentication using registered IP address and port
You can view further details and register for service here : Skype for SIP Beta Open
If you liked this tutorial ,don't hesitate to buy me a Cup of Coffee today !
(2 votes, average: 4.50 out of 5)
February 25th, 2010 at 5:28 am
Thanks for the post. Kindly can you please create a video tutorial also. I would be grateful.