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denuviker
Intermediate
Joined: June 27 2006
Location: United States
Posts: 2
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| Posted: June 27 2006 at 5:42pm | IP Logged
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Hi,
We had a couple questions:
Are hung calls detected?
Can a sip user transfer a landline caller to an external number?
Is it "hands-off" the media stream if both endpoints don't need NAT traversal?
Sipura phones supported?
How many servers (P4 3ghz, 1 GB RAM) do we need to support 2,000 users?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 28 2006 at 10:24am | IP Logged
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Hi denuviker,
You>>>
Are hung calls detected?
<<<Support
Yes they are. For example: If two IP phones are in a call and we unplug their network connections, the media proxy for the call will detect that no RTP data is flowing and terminate the media session. This "RTP activity time" can be specified via GUI configuration parameters.
You>>>
Can a sip user transfer a landline caller to an external number?
<<<Support
No. Note that the capability you speak of is not a limitation of our products. The limitation is in how SIP and the PSTN interoperate.
You>>>
Is it "hands-off" the media stream if both endpoints don't need NAT traversal?
<<<Support
Yes, exactly.
You>>>
Sipura phones supported?
<<<Support
You bet. We just completed our latest interoperability testing using the Linksys-Sipura 9xx phones. We like them. They work well.
Take a look at the following version history to see what SIP devices we just tested:
Centrex Proxy Version history
You>>>
How many servers (P4 3ghz, 1 GB RAM) do we need to support 2,000 users?
<<<Support
Unfortunately we do not yet publish load capacity data for the servers. The primary reason is that it varies widely based on operating system, underlying host hardware, network drivers, NIC interrupt latency, and bandwidth of the network connection.
We know customers want this data and we are working to develop some form of normalized data. As a rough estimate, we will assume the following: Operating system of Windows XP Pro or 2003 Server and a single 1 megabit Ethernet network connection. The max number of concurrent calls your deployment can support is limited to the max number of media sessions needed to support NAT users (Note: We are not considering calls that do not require proxied media).
Assuming all your SIP endpoints use 20Ms G729 media for full duplex media proxied voice only, a single call consumes around 2200 bytes per second. Assume 10 bits per Ethernet frame, the network call capacity is around 4500 concurrent calls. For Ethernet collisions and network errors, assume 70% of this figure. This brings us to 3100 calls per megabit Ethernet connection.
In reality, the interrupt service time for the network adaptor will be the limiting factor. The host machine can only process a certain number of interrupts from the NIC. If the host cannot service the NIC interrupts fast enough, the NIC will probably start experiencing receiver data overruns and toss the received data out. Transmitter overruns will also occur.
As far as we know, no generic host machine will be able to handle 3100 active calls (1 VOIP Media Proxy). This would mean an interrupt service rate of 2 * 155k interrupts per second. This assume NIC receive and transmit interrupts are being generated by the NIC.
So what is the real answer to your 2000 user question? We suggest 2 media proxies running on their own dedicated host machine.
What you can do is create a “media proxy farm” and size your deployment for 1000 concurrent calls per media proxy. As the number of active sessions grows, just add another media proxy. The Centrex proxy will load share all your media proxies.
If you connect your media proxies to gigabit Ethernet, then you should be able to scale your media proxy farm to around 30 to 45 media proxies – give or take.
Repost as needed,
Support
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denuviker
Intermediate
Joined: June 27 2006
Location: United States
Posts: 2
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| Posted: June 30 2006 at 10:13am | IP Logged
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Great responses thank you. I wonder if you might be able to go into more detail on the hung calls:
support wrote:
Hi denuviker,
You>>>
Are hung calls detected?
<<<Support
Yes they are. For example: If two IP phones are in a call and we unplug their network connections, the media proxy for the call will detect that no RTP data is flowing and terminate the media session. This "RTP activity time" can be specified via GUI configuration parameters.
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What would happen if both sides of the call are not behind a NAT, and the media isn't being proxied, so there's no way to monitor the RTP data. How would a disconnect be detected?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 30 2006 at 12:20pm | IP Logged
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Hi denuviker,
You >>>
What would happen if both sides of the call are not behind a NAT, and the media isn't being proxied, so there's no way to monitor the RTP data. How would a disconnect be detected?
<<< Support
In the case for calls not having their media proxied and both call endpoints puke, there's not much you can do. I have a sneaking suspicion why you are asking this question :)
More Info:
Internally the Centrex (SIP) proxy knows if a call has lasted more that a "maximum" amount of time. This max time is configurable and basically sets the max duration that internal call state information will be managed. The settings I am referring to are in the media proxy support config dialog of the Centrex proxy. They are labeled "Call history support".
Unfortunately for the current release, the SIP proxy just removes "stale" or dead calls from its state info after the set time. I will look into it further. It seems to me that we need additional functionality here. I know we are working on adding call detail records and other such functionality.
I assume you are asking about "hung" calls due to you wanting to bill customers accurately. If you can relay to us some additional specifics regarding your deployment, then we can be of more help.
One nice thing about LanScape - When new functionality is identified via our support group and this forum, we pass the information to our developers and they are quick to respond.
Repost as needed,
Support
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LanScape Support Forum -> SIP Proxy and Media Proxy - Pre-Sales Technical Support
Topic: Deploying ISTP of 5000 users - Questions
