Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Who should I go with that would guarantee me quality service just like an analog line? _________________________________________________________________ See what you?re getting into?before you go there http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507
There is a strong possibility that the problem is on your side. Are you using a cable or dsl? What are your download and upload speeds? Are you doing any kind of traffic shaping? You will not get a guarantee of QoS from any provider. They cannot control what is happening on your end or what happens on the public internet. Maybe if you put in a point to point to the provider, then they might consider it. If you are seriously considering doing business using VoIP, then I would reconsider unless your internet provider is providing the VoIP service and they observe QoS on their equipment. Otherwise, you can never be sure what the quality will be at any given time. Weigh the saving against the cost of dropped or garbled calls. Thanks, Steve John Meksavan wrote:> Asterisk Users, > > I recently ran into some problems with the quality of service with > Teliax. This occurred on August 1, 2007 with a dropped outbound call, > audio quality isse on the callee side- not hearing me well on callee > side, and sending DTMF tones (configured for RFC2833). Am I the only > Teliax customer having this problem? > > It seems like when I am ready to go live with my Asterisk PBX System, > I run into quality of service issues with the SIP provider. Who should > I go with that would guarantee me quality service just like an analog > line? > > _________________________________________________________________ > See what you?re getting into?before you go there > http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507 > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
At 09:23 AM 8/2/2007, you wrote:> I recently ran into some problems with the quality of service with > Teliax. This occurred on August 1, 2007 with a dropped outbound > call, audio quality isse on the callee side- not hearing me well on > callee side, and sending DTMF tones (configured for RFC2833). Am I > the only Teliax customer having this problem?Teliax has been quite good. I was having problems the last 2 days and they confirmed that they are working on fixing something. I've been using IP for all my outgoing calls for the last couple of years and other than being ripped off by a couple of vendors and the occasional connection problem it's saved me large amounts of money, more than what I lost when the 2 providers refused to return my deposits and then went under, but I do have ways to get dial tone on my POTS lines for those times when it all goes to heck. Ira
John Meksavan wrote:> Asterisk Users, > > I recently ran into some problems with the quality of service with > Teliax. This occurred on August 1, 2007 with a dropped outbound call, > audio quality isse on the callee side- not hearing me well on callee > side, and sending DTMF tones (configured for RFC2833). Am I the only > Teliax customer having this problem? > > It seems like when I am ready to go live with my Asterisk PBX System, I > run into quality of service issues with the SIP provider. Who should I > go with that would guarantee me quality service just like an analog line?If you want service to be as reliable as the PSTN then you have to use the PSTN. I feel that sending calls over the Internet is just silly if you want as close to %100 uptime as you can. My customers use PRIs with VoIPoInternet as a failover in case the PRI goes down or all channels are in use on the PRI. I am not saying that VoIP is unreliable. It is very reliable -- when you control the lines and routers between you and the PSTN. I'm saying that the Internet is not reliable. My customers route calls over point to point T-1s all the time with no issues. Teliax seems to one of the better ITSPs.
On 8/2/07, John Meksavan wrote:> Asterisk Users, > > I recently ran into some problems with the quality of service with Teliax. > This occurred on August 1, 2007 with a dropped outbound call, audio > quality isse on the callee side- not hearing me well on callee side, and > sending DTMF tones (configured for RFC2833). Am I the only Teliax > customer having this problem?ditto here this week, random breaks in audio, garbled voice etc. My softphones dialing in from outside had no audio issues. Others on teliax forums suggested I switch to SIP since iax2 is aggressively evolving and teliax equipment is experiencing some incompatibilities with recent * iax releases. I changed codecs from gsm to ulaw, voice quality improved but same random breaks.> It seems like when I am ready to go live with my Asterisk PBX System, I > run into quality of service issues with the SIP provider.Consider having some fall back options from alternate providers since it doesn't cost a whole lot to keep an active account.> Who should I go with that would guarantee me quality service just like > an analog line?I have heard that there is no such thing unless your provider & you have a dedicated, or at least highly reliable, circuit between the two of you : http://en.wikipedia.org/wiki/User_Datagram_Protocol "UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, at least for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets..." One of the reasons Time Warner, Armstrong, Cox and other cable broadband guys are able to offer fairly reliable voip service is that they control the pipes between their VoIP proxies and their end users. It is also the reason vonage, teliax and other 3rd party vendors have more issues. I used broadvox a few years ago, if the callee answered before the caller had heard a ring the line went dead :-) -baji. --
On Aug 2 2007, John Meksavan wrote:>Asterisk Users, > > I recently ran into some problems with the quality of service with > Teliax. > This occurred on August 1, 2007 with a dropped outbound call, audio > quality isse on the callee side- not hearing me well on callee side, and > sending DTMF tones (configured for RFC2833). Am I the only Teliax > customer having this problem? > > It seems like when I am ready to go live with my Asterisk PBX System, I >run into quality of service issues with the SIP provider. Who should I go >with that would guarantee me quality service just like an analog line?VoIP is susceptible to packet delivery problems anywhere between your PBX and your SIP provider's PRI lines/termination point. If you have direct SIP PBX to SIP PBX calls, then your problems can be anywhere on the Internet path between the sites. The only workaround that I know of is having your ISP be your SIP provider, so that your SIP packets only cross your ISP's own network to its termination point, and do not cross the public Internet. This way QoS can work from your office to your ISP's office to make sure you maintain reliability. I have not personally used iTEL-ip's 'iTEL Voice Service', but others have said, as do their own notes that their network QoS is effective at maintaining call quality. When I contacted them, their pricing for a 'QoS private IP backbone for voice and data' was $618/month for a full 1.5mbps T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending on contract length. Per minute rates were $.03. When I ran the numbers, it appeared that a regular full T1 + a regular full PRI would be only slightly more. A major tradeoff comes in the physical location flexibility you get with SIP over traditional phone lines in the case you need to move an office (although physically moving the phones to a non iTEL-ip data line would mean you're not getting their Qos). iTEL-ip's 'iTEL Voice Service' http://www.itelconnect.com/default.aspx?type=t§ion=iTEL-ipVoiceService&selection=16 http://wiki.pbxnsip.com/index.php/ITEL-ip -hk
You know the problem is that most consumers think that it is possible to get the best and the most reliable for almost nothing. They go out with this expectation and get the cheapest, then when it bites them a few times, they scream "why me". ---------- Original Message ---------------------------------- From: SIP <sip at arcdiv.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com> Date: Sun, 05 Aug 2007 19:49:40 -0400>Worthless comes in many forms, Doug. If you're talking specifically >about the monetisation of hardware/effort, then it may indeed be >worthless by the simple fact that the cost may outweigh the net gains in >profits gained from the purchasing, configuration, and deployment. > >Businesses are about making money first and foremost. If the amount of >time and money put into a particular project outweighs the money you get >in return, it's a bad business decision.________________________________________________________________ Sent via the WebMail system at rockynet.com