Depending on you bandwidth, you might not need QoS. Priority could be enough.
In you sip.conf (if you use SIP), place a tos value: [general] tos = 0x10 ; low delay or tos = 0x46 ; DiffServ Premium (EF: Expedited Forward) Remark: for un unknown reason, tos=lowdelay doesn't work anymore on my asterisk (v1.0.7), but was working in the past. I replaced it by 0x10 (hex value of lowdelay). Most of the routers support PFIFO (FIFO with priority), which means that low delay flagged packet will be sent in priority. I haven't tested the 0x46 value yet. Routers must be configured for DiffServ values, while ToS is by default. But the low delay TOS bit is also set within the 0x46 value. If a router treat the the DiffServ byte as TOS, it should be sent with priority as well (to be validated). If you want to check what priority is set inside your packets, you might use Ethereal. You might see either UDP or RTP packets, depending on the RTP ports that are used. In the branch "Internet Protocol", you will find the TOS/DiffServ decode, named "Type of service" or "Differential services Field". The TOS low delay bit is the 5th, and should be 1. If you have a low bandwidth connection (e.g. 600/100), you might have a new problem if you are using TOS as low delay. Voice will be good, but data will stall. QoS won't resolve it, because big packets take too much time to travell. The only way to share bandwidth for voice and data, on low bandwidth lines, is to fragment the data. An MTU of 700 is quite good, but you have to assume about 15% of bandwidth loss, because of twice more overheads on big packets. Allthough, a 1200/200 kbps line usually doesn't require such tricks. Remark about Grandstream: If you are using a GS device, you must know that QoS is buggy, and will have no effect at all. You must upgrade to the beta version of the firmware, which is OK. Therefore, GS recommands a QoS value of 48 (whithout "0x" on a GS device). This is a DiffServ value, which does not set the los delay TOS bit. Cisco recommands 46, which does. Jean-Christophe chawki hammoud a écrit : >--- Matt Riddell <[EMAIL PROTECTED]> wrote: > > > > >>Assuming your provider completely ignores QOS, it is >>still not a >>complete waste of time. >> >>If for example you have 5 people on the LAN, 4 >>uploading files to a >>remote server and 1 trying to make a phone call. >> >> > >My ISP has the internet connection set-up where 8 >people share the bandwidth. Would the script still >help boost my voip calls? > > > > > >__________________________________ >Yahoo! Mail Mobile >Take Yahoo! Mail with you! Check email on your mobile phone. >http://mobile.yahoo.com/learn/mail >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users