www.ralden.com - Technical Support

Subject	Text
Call Detail Records	You can access your account balance and call detail records following the instructions on page: http://www.ralden.com/Call%20Detail%20Records/
Codec Support	Our system supports G.711 (both ulaw/alaw) and iLBC. We are working on getting get G.722, G.729a and G.723 support but don't have them yet. G.711 is the most common codec and delivers very good audio quality, but it requires as much as 87.2 Kbps of bandwidth to perform perfectly. For low-bandwidth situations iLBC is of excellent quality; it consumes less than 16 kilobits per second of bandwidth. But, it is not widely supported in telephony hardware adaptors; but there are some, such as the Grandstream product line. When you configure your SIP endpoint you may be asked to select a codec or choose a preference or rank ordering of codecs for negotiation. For our service you should select G.711 or G.711u as your preferred codec if you can afford the bandwidth and iLBC if you must use a low bandwidth codec. Some Windows softphones use the Microsoft g723.1 codec and often the product documentation does not mention any other codecs. However, often these systems will negotiate G.711 (which is built into most computer systems), and will work with our system quite well. Two softphones which support the iLBC codec are X-Lite from www.xten.com and SJphone from www.sjlabs.com. Both have been tested with our service and work. X-Lite has a ghastly user interface that can be difficult to get working, but it is also a softphone many use as a baseline. SJphone is easier to get to work but has obvious flaws (for example, it will “crash” if you press the dial button with no number entered). Other than that drawback, it works quite well, and it favors the iLBC codec “out of the box” (X-Lite requires significant customization just to say you prefer iLBC).
Dialing Patterns	All numbers should be dialed in e.164 format. That is, the country code followed by the remainder of the number. Do not "dial" any sort of prefixes such as 00 or 001, etc. which might be common for a conventional phone system. Note that it is often the case that what appears after the country code is a "city" code which sometimes starts with a 0; it is often necessary to remove this leading 0 when a call arrives through an "international" gateway. For example, a number in London, UK might be written: 0207 629 1234 but when dialed with the UK country code 44 it must be: 44 207 629 1234 This pattern is common in many countries besides the UK. When written, and in many computer systems, country codes are prefixed with a + sign. E.g., +44 207 629 1234. If you are using a software phone you must omit the + sign from your dial string as well as any other non-numeric characters, such as ( ) -. The string that reaches us must be a pure string of digits. I.e., No: +44 (207) 629-1234 Yes: 442076291234
Gateway Prefixes	You can call other VoIP networks by dialing a special prefix. We charge $0.012/minute for these calls. To dial Free World Dialup numbers use prefix 000393 To dial Gosiptel numbers use prefix 000467
General SIP Endpoint Configuration	To configure each device (soft phone, hard phone, or other SIP equipment on your end) to use our service you will need several items of information which we provide you when an account is established. 1. The DNS name of our sip gateway (this is sip.ralden.com). 2. For each device, two SIP endpoint identifiers: The first we call a “dialable” ID which looks like a telephone number that starts with 87800 (and indeed can be used to place a call to a SIP endpoint on our network); the second is an “authorization” ID which is accompanied by; 3. an authorization password. Your system will perform two separate actions that involve this information. The first is to "register". Registration is performed on a regular basis and allows our system to know how to contact you if we need to send a call to you. A successful registration also informs you that you have your credentials (name and password) set up correctly. If you mistype your name or password or otherwise make a mistake, registration will fail. You do not have to configure your end to register if you do not want to; registration is optional. The second step is to make a call. In SIP terminology this involves the "invite" message. For every invite we will challenge your end for your correct authorization ID and password. Assuming registration worked earlier, these should already be in the correct form. Many SIP clients (hardware and software) have a number of confusing options and use different terminology. SIP makes a distinction between a "user" identifier and a "logon" or "authorization" identifier. In many systems the user identifier looks like a telephone number because it determines how the endpoint is addressed in terms of dialed numbers. The authorization identifier is more important; it pairs with a password and establishes your authority to access a particular service. In our system the “authorization” ID we assign you can be used for both purposes; so, when in doubt, try that. You only need to configure the “dialable” ID if you want to place “on network” calls to your endpoint. In our call accounting records all calls are noted by “authorization” ID and it is the “authorization” ID which can have an independent credit limit. Again, the “dialable” ID and the “authorization” ID come together in related pairs but you do not have to use or configure the “dialable” ID unless you want to receive inbound calls using that number assignment from us. Generally the correct place to configure the password is fairly obvious. There are another set of options which are potentially confusing. A SIP identifier is like an email address, so technically there is an @something.com on the end of it. However, you generally never enter that with the "user" ID part of the SIP ID; rather it will be entered elsewhere. Often it will be referred to as the "domain" or sometimes the "gatekeeper" or the "proxy". In all cases these parameters expect to see an IP address or a DNS name. For our service, when in doubt, use sip.ralden.com. Your system may refer to a proxy, an "outbound" proxy and possibly an "alternate" proxy. When in doubt, use sip.ralden.com for the "proxy" and leave the others blank. If your endpoint is behind a NAT device you should still not need to do any special configuration (such as entering a parameter for a "stun" server). Our system works fine for endpoints behind NAT devices which are only making outbound calls. For inbound calls there are some issues to consider which we will review if you are an inbound customer.
Test Numbers	We have a gateway to Free World Dialup (FWD) so that you can call their test numbers to help debug your SIP configuration without having to call your friends in the middle of the night :) Dial 000393 + the FWD number; here are some useful test numbers: 000393613 Echo Test (test quality/latency of your connection). This is the most useful number; whatever you say is repeated from the far end and you can get a good idea of current audio quality. 000393612 Automated Time Announcement (reads out current United States EST, UTC-5). 000393614 Digital Milliwatt test (See http://www.atis.org/tg2k/_digital_milliwatt.html for an explanation). 000393615 Voxeo Tone test. Press 1-7 to select one of seven different test tones; press # to stop playing a tone and choose a different tone. You can, of course, dial other FWD subscribers by using the 000393 prefix. Calls over 5 minutes are charged $ 0.01 per minute.