Using ZipDX Wideband
Notes on connecting to the VUC Friday call on ZipDX
Rev 5 Nov 2013
Connecting via PSTN: Dial +1 646 475 2098. Since the discussion is about VoIP anyway, our expectation is that most participants will connect using SIP, which is usually less expensive.
Connecting via SIP: Use this SIP URI: email@example.com. The rest of this document provides some hints regarding successful (and unsuccessful) SIP connections. If you are having trouble connecting via SIP, please read the following. Unfortunately, SIP can get somewhat complex and you may need technical knowledge to work through any issues.
Wideband: The ZipDX conference bridge supports wideband audio. Most of the “regular” participants connect in wideband and the group encourages everybody to connect that way. Speech is clearer, it is easier to recognize voices, there is less repetition, it isn’t fatiguing, and it just plain sounds better.
The ONLY wideband codec currently supported by ZipDX is G.722 (NOT G.722.1 or G.722.2, nor Speex or SILK or iSAC or iPCM-wb). Make sure that your SIP endpoint is configured with the G.722 codec enabled and that it is at the top of the codec priority list. (Even if it isn’t at the top of the priority list, ZipDX will TRY to select it if your endpoint offers it.) Opus support is experimental.
You can also connect to ZipDX in NARROWBAND, using either G.711 a-law or mu-law (PCM-A or PCM-U). These are the ONLY narrowband codecs supported by ZipDX (NOT G.729 or any others).
When you connect to the ZipDX conference, the system will announce either “Now joining the bridge” or “Now joining in wideband.” You’ll hear the second version if you are connected with G.722.
SIP Signaling & Configuration: ZipDX uses SIP standard signaling. Whatever you use to connect should support the SIP standard. Our experience is that many systems claiming SIP support are not fully compliant with the standard. This is especially true of new or “beta” devices or software releases.
You should set your configuration to use “RFC2833” to send DTMF tones (rather than “SIP INFO” or “in-band”).
ZipDX uses some SIP messages that may give trouble to non-compliant SIP implementations. Specifically, ZipDX will send INVITE messages subsequent to the initial INVITE used to establish the connection. These are called “re-INVITEs.” Your SIP client must support re-INVITE in order to connect (and stay connected) to ZipDX.
ZipDX also sends UPDATE messages. Even if your SIP client does not take any action on the UPDATE message, it must at a minimum send a response (saying that UPDATE is not supported, for example). Your SIP client should not get “confused” when it receives an UPDATE message.
Initial Connection: To get connected initially, you have to send an INVITE message to ZipDX; that has to be successfully received at ZipDX, and it will then send back TRYING, RINGING, and OK messages. If you are not successful connecting, it is most likely because:
* You have not properly entered the SIP URI (firstname.lastname@example.org)
* Your SIP client is not properly resolving the “login.zipdx.com” domain. There may be a problem with the DNS that it is using (or it may not have DNS properly configured).
* You have a “proxy” (including, perhaps, a PBX) between your SIP client and the Internet that is blocking the messages.
* There is an “Application Level Gateway” or some other intermediate device between your client and the Internet that is modifying or blocking the messages to or from ZipDX.
* Your client is not properly connected to the Internet.
* There is no compatible codec for ZipDX to use. (Only G.722 and G.711 PCMA/PCMU are supported.)
* You have offered a media stream that ZipDX does not support. (Turn off “video” if you have that enabled.)
* The conference is not active. You will hear a recording telling you that the “code is not recognized.” Call back at the scheduled time.
If you cannot get connected, check the items listed above. See also “Testing” below.
Disconnects: If you get disconnected after successfully connecting, it is probably because your client is not properly handling re-INVITEs or UPDATEs.
ZipDX constantly monitors the jitter associated with the packets it receives. If it needs to make an adjustment to its jitter buffer, it will do so, and then send a re-INVITE. Your SIP client must respond properly, or the call will disconnect.
ZipDX also periodically sends an UPDATE message to make sure that your client is still active. If it does not get a response (even one that says UPDATE is not supported), it will terminate your connection.
One-Way or No Audio: When you first join the conference, you will typically be muted. (You’ll hear an announcement, “Now muted. Press star-six to un-mute.”) You should be able to hear the conference audio. If you wish to speak, dial *6; you’ll hear “You are now un-muted.” If you do not hear this announcement, the most likely cause is a problem in your DTMF (RFC2833) configuration.
If you have other audio problems, try placing your call on HOLD (if your phone offers that option) and then retrieving it. If that fixes the problem, it probably means that your SIP client is not properly handling the re-INVITEs that ZipDX sends.
Other potential causes of audio problems include:
* A firewall is blocking the “RTP” packets that carry the audio.
* Your client is operating in “NAT” mode. This means that it is waiting for ZipDX to send audio packets before it will send its own packets. However, ZipDX does the same thing – it will not send audio packets until it receives them from you. Disable NAT in your configuration.
* There is some other problem with your Internet connection.
Also see “Testing” below.
HardPhones: We have had success connecting the following SIP telephones to ZipDX: Aastra, AudioCodes, Avaya, Cisco, Gigaset, Grandstream, Konftel, Polycom, snom. There are versions of all of these that support G.722. However, there are many versions of firmware. If you are having trouble connecting, check for a newer version of firmware. Generally, the more recent production firmware releases are better than their predecessors.
The Gigaset does not allow SIP URI dialing. On the Gigaset, you may be able to connect by dialing 2371999#9, and then entering code 200901# at the prompt. However, the reliability of this connection mechanism is not high. You can also configure the Gigaset to register with ZipDX; this is a more involved process.
SoftPhones: ZipDX works well with CounterPath’s eyeBeam product, provided it is properly configured. Not all versions of eyeBeam support G.722. There are numerous other softphones available; our experience is that many have serious flaws in their implementation. Software updates are frequent so your mileage may vary.
If you have trouble connecting and can get support from your softphone supplier, contact them. If you are using a “free” softphone, remember that often “you get what you pay for.”
As of November 2009, we are not aware of any softphone for the Mac that supports G.722.
Asterisk: You can connect to ZipDX through Asterisk. Bear in mind that there are MANY versions of Asterisk and numerous patches and options that can be installed. Generally, version 1.6 fully supports G.722, but version 1.4 does not.
Web dialer: Thanks to Tim Panton and PhoneFromHere.com you can call the ZipDX (g722) bridge with the Phone from Here Widget
We are aware of the following Asterisk-related issues:
* There is a problem with sip session timers in asterisk 188.8.131.52. We managed to work around it by setting session-timers=refuse in sip.conf
* If you have Video support enabled, disable it for ZipDX calls.
Testing: You can test your setup at any time by calling sip:email@example.com. ZipDX will play a recording, including an announcement indicating if you are connected in wideband or narrowband. Also, once the recording is playing, you can press “2” to force ZipDX to issue a re-INVITE message, so you can verify that your client responds properly. (It should not disconnect; you’ll hear a message and the recording will continue to play.)
Trouble Reporting: If you are having trouble connecting, you can send an E-mail to firstname.lastname@example.org and we will attempt to help you, time permitting. All of the following will be useful to us in diagnosing the problem:
* The date and time(s) (including time zone) of your call(s).
* A description of what did or did not work (what you heard, what other indications you saw on your telephone).
* The type of SIP client (hard or soft phone, plus firmware/software version information) you are using.
* Something to help us identify your origination information (your public IP address, contents of your FROM header and/or Display Name).
* Information about your system set-up (“Polycom IP-650 connected to an Asterisk 1.6 server, configured to access the VUC SIP URI when I dial extension 9955…”).
* SIP trace or packet capture showing the failure.
The more you can send us, the more likely we will be able to help.