On Wed, Nov 3, 2010 at 1:17 PM, Tom Chadwin <[email protected]> wrote: >> it may also be that the Burst >> network traffic is causing some of the SIP packets to be lost >> going to your handsets.. > > How can one mitigate against this?
The network traffic itself is not a problem; it's the rate of call setups per second... When you configure a page group using SIP and MeetMe you're asking Asterisk to setup X (in this case 20) calls to SIP endpoints at the same time. Assuming Asterisk can even queue things up quickly enough you're essentially asking it to do somewhere in the neighborhood of 20 call setups per second, which is quite high for Asterisk on this hardware. Then the RTP starts flowing and things go downhill from there... >> or perhaps go to sip >> TCP instead of UDP if your version of asterisk and the >> handsets will support it.. this would give a lot better >> chance of the packets making it... > > I am using whatever the default Astlinux install uses. Does 1.4.x handle SIP > over TCP? Does TCP by definition increase latency? Asterisk 1.4 doesn't support SIP over TCP (1.6 and later do). However, this is probably only going to make things worse as TCP has higher overhead and requires more resources than UDP. UDP doesn't guarantee delivery at OSI layer four but that doesn't mean delivery guarantee/retransmissions can't be implemented at other layers of the model, as has been done with SIP. SIP has mechanisms to retransmit UDP that generally work absent a few tricky scenarios that probably don't apply here. Asterisk 1.8 would really help you because it supports multicast paging and from what I've seen you're using Snom phones that support multicast. This greatly reduces the system load for two reasons: 1) Asterisk doesn't have to establish SIP sessions to each member of the page group. 2) Asterisk sends out one copy of the RTP to an RTP multicast address instead of one copy of RTP (unicast) to each station. With multicast RTP paging Asterisk and your net5501 don't care if you have one phone or 100 phones in your page group. It's a very nice feature and I'm glad I was involved in getting it implemented in Asterisk :). >> The other thing I do is use only ulaw for handsets connected >> to the LAN.. >> and all my voice files are also in ulaw.. > > I've tried to ensure that this is the case, but I find working with multiple > codecs a bit of a black art. We are alaw over here in the UK, though I guess > I could use ulaw in Asterisk, and let the ISDN hardware handle that > conversion. I use GSM across the IAX trunks between offices, since they only > have max 0.8Mbps, so bandwidth is an issue. > > I leave ulaw in there to try to avoid unnecessary transcoding, but I am > rather taking it on trust that this is working. Also, where does Asterisk > native sound format (is it .sln?) come into this consideration? > You're right to describe codec transcoding and negotiation as a black art; it can get tricky quickly. I've seen a lot of setups where codec negotiation has run amok and calls end up getting transcoded and re transcoded so many times the call quality is terrible... If you're in the UK stick with alaw, there's probably not much advantage for you to use ulaw like us yanks :). If you're using multiple codecs and playing files from disk Asterisk will look for the "easiest" file to playback. If you're using applications (MeetMe) that require slinear (Asterisk's native format) you'll be transcoding and now there is a big difference between alaw and GSM. It's complicated. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ------------------------------------------------------------------------------ The Next 800 Companies to Lead America's Growth: New Video Whitepaper David G. Thomson, author of the best-selling book "Blueprint to a Billion" shares his insights and actions to help propel your business during the next growth cycle. Listen Now! http://p.sf.net/sfu/SAP-dev2dev _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
