-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I have an installation where I'll have a site to site data DS1 for use between two corporate offices. We'll have one asterisk server at each office. I'd like to be able to route calls over the 24 channels on that DS1 between the offices, instead of over the voiceT at each location to maximize savings on interoffice calls. An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? Thanks for your help, Jonathan -----BEGIN PGP SIGNATURE----- Version: PGP Universal 2.0.6 iQEVAwUBRKJtFpJhYmFK+jfsAQh/tggAiqCqlefhEyAuIcshX5AaMGx3flVdHn5C mh1TY5i/Z8tf4LBEh+TuXvUFGNXvnPn12nrEwkF8s4HOUcDwVhAXI5XlA7WZFT83 H3UGoK7RGaitirWHDKFEfa3+BlWpL8eclsdItGx0FPHtdQeRCxq2ba1gtKszpaHC KgApM9ExYVwEPFcwbYoK2m0pvofuiYNYxw/yN7ZkIooM1oWTP8NFjGuysrb2FW2J 8odHb+J8ySmhmHQFWZ+XVHnkOTckp+feaKUuCohsffBxBm5mPrdXpQMwnCCR5yhz bhoAaveMPJz7gcSIgXTAMyZtO4m8U3/zht443S1J/MTD30seL8goPg==r6vs -----END PGP SIGNATURE-----
On Wednesday 28 June 2006 08:48, Jonathan Miller wrote:> An alternative is to put a router and switch at each end and extend a data > network to the other site for SIP traffic. Would that result in better > quality calls?If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true "point to point" -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work.> What configuration areas are there to be set and how are they diffent than > just a standard PRI, which I have working now?If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A.
Hi Jonathan -> I have an installation where I'll have a site to site data DS1 for use between > two corporate offices. We'll have one asterisk server at each office. I'd > like to be able to route calls over the 24 channels on that DS1 between the > offices, instead of over the voiceT at each location to maximize savings on > interoffice calls. > > An alternative is to put a router and switch at each end and extend a data > network to the other site for SIP traffic. Would that result in better > quality calls?You'll get better quality calls by using the 24 channels of the T1 directly as voice channels. They'll be high-quality ulaw calls, but you'll be limited to 23 simultaneous calls over the link. If that's OK with the client, I'd go that route. You can avoid QoS setup and jitterbuffer configuration. On the other hand, if they want more simultaneous calls than that over this link, you could use it as a data T1, and use g729, and you could fit a LOT more calls over this link. They'll be lower quality just because of the g729 codec, and you also have to deal with QoS and jitterbuffer. - Noah