jugg
join:2007-10-10
| what voip providers don't proxy media sessions?
Hello,
Does anyone know of a VOIP provider that does not proxy the media session, but instead allows the RTP session to be initiated directly to the termination center?
I'm using callcentric, and all voice traffic is routed through their servers in New York. Because I am currently outside of the US, calling any number outside of the US is subject to many failure points along the network path and usually results in very poor quality. If I'm in Asia and place a call somewhere in Asia, there is no need for my voice traffic to be routed through New York. Rather it should be routed directly to the termination point or regional location within Asia.
Also, for those who will immediately reply with "do you have an Outbound Proxy configured?", please do some more research... "Outbound Proxy" (at least on SPA devices which I'm using) has nothing to do with the RTP session - it only controls how SIP messages during the call are handled. Besides which there isn't anything the client side can do to control what IP address the VOIP provider tells the client to establish the RTP session with.
But, back to my question - what VOIP providers route directly (or at least regionally) to the termination center, rather than through their central home servers?
Bonus question: Are there any Asia based VOIP providers that have an English language web account interface?
regards,
chris |
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mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| said by jugg  :But, back to my question - what VOIP providers route directly (or at least regionally) to the termination center, rather than through their central home servers? To name a few. AFAIK, most VoSPs, i.e FWD, SIPPhone, etc., no longer route RTP media traffics through their server once a P2P SIP call gets established.
Bonus question: Are there any Asia based VOIP providers that have an English language web account interface? How about pfingo?
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anon99
@cox.net
| reply to jugg
I use CallWithUs for my toll free calls because they do not proxy the RTP traffic(at least with my setup anyways). I don't have an account with them for paid calls but I would think it would be the same.
I'm sure others can give you some more ideas.. |
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sokhapkin
Premium
join:2003-05-08
Somerset, NJ
| reply to jugg
CallWithUs server auto detects NATed SIP clients. If the client is on a public IP address (or STUN is working properly), then direct audio path between call termination GW and SIP client is set up. Otherwise audio is proxied by one of CWU servers closest to SIP client geographical location (identified by SIP client external IP address).
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»www.callwithus.com |
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no_one
@qwest.net | reply to jugg
Question? Wouldn't the carrier used by the VOIP provider still need a termination center near where you wanted? So even if not proxied the final carrier may not have a termination point nearby? |
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dicodread
join:2008-11-16
barbados
 ·Callcentric
·callwithus
| reply to jugg
The only provider I know that did that proxy thing was inphonex and I found it added a delay to the call.
I use CWUS and callcentric and both allows my devices to connect directly to the gateway provided I am able to use the required codec. I realised this because I had to change the name of my g.729a codec to g.729 because some international gateways didnt recognize g.729a/g.729ab. The only reason that became a problem was because we were talking direct. |
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jugg
join:2007-10-10
| reply to jugg
Just to clarify, I'm talking about PSTN terminated calls.
When a call is placed, the IP portion has to terminate somewhere to be connected out to PSTN. I'd like the IP portion of that call to be routed as close to that termination point as possible.
Callcentric routes all PSTN terminated calls through their New York server, regardless of the call destination.
Many providers of course allow IP to IP calls to have their RTP traffic routed directly, but that is not what I'm asking about.
I'm looking for a provider that routes the RTP traffic as near to the PSTN termination point as possible - at least to various regional centers. |
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engineerdan
join:2006-12-07
Manassas, VA
·Callcentric
| said by jugg :Just to clarify, I'm talking about PSTN terminated calls. When a call is placed, the IP portion has to terminate somewhere to be connected out to PSTN. I'd like the IP portion of that call to be routed as close to that termination point as possible. CallWithUs and FlowRoute both hand off the voice media directly to their carrier(s). This may or may not be near the termination point.
CallWithUs offers a unique service that permits end users to use the CallWithUs on-line "call simulator" to see a list of carriers (or, more accurately, carrier code numbers) that might be used to place a particular call. The list is sorted by cost with the least expensive carrier (the default route) listed first. You may also select a specific carrier by prepending the carrier code number to the dialed telephone number.
In the case of IP PBX users or ATA users who are comfortable tinkering with their dial plans, this feature can be used to select from among as many as ten carriers.
Although CallWithUs is generally quite reliable by default, I recently had some trouble terminating to a telephone number located in rural Minnesota. After a handful of test calls using various CallWithUs carriers, I found two carriers that worked quite well.
Out of curiosity, I ran a traceroute to each carrier's IP. One is based in New York; the other in Canada. he New York carrier was less expensive, so I modified my Asterisk dial plan to select that particular carrier whenever that rural Minnesota exchange is dialed. In this case, my best results were achieved by selecting a carrier that terminated many hundreds of miles from Minnesota. |
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CallC
@telengy.net
| reply to jugg
Information notice:
Callcentric Wholesale, Inc. (a parent of Callcentric) owns and operates its own TDM/SS7 network interconnected with major (as well as small ones) carriers, networks and voice exchanges - over 800 interconnections, nationally and internationally directly from its switch in NY headquarters. We guarantee an optimal voice (media) path to most domestic and international destinations.
Callcentric Wholesale is also an ARIN registered ISP, operating its own IP network blocks and exchanging BGP-4 routing information with most ISP backbones (also nationally and internationally) via its own managed fiber optical network thus guarantying the best IP packet delivery to virtually anywhere in the world (we monitor such parameters as round-trip time, packet loss and number of hops to most international destinations and adjust our routing according, if/when required).
Our average and sustained round-trip time to:
Hong Kong is 225ms while we're reaching Tokyo (Japan) in under 175ms. Sydney (Australia) is under 230ms; almost all European destinations are under 150ms (London - 75ms; Paris - 85ms, Berlin - 90ms); Israel - 120ms, Jordan - 120ms, Egypt - 120ms, Saudi Arabia - 120ms, Pakistan - 125ms, Kuwait - 140ms, most cities in India - under 150ms, Bangladesh - 200ms (Bangladeshi national fiber backbones are quite overloaded with no signs of expansion; satellite link under 850ms); Mexico - 115ms, Brazil - 135ms [Sao Paulo];
USA from NY to West coast - under 80ms, to Miami - 35ms; Canada to Toronto - 25ms, to Vancouver - 85ms (at worst, most of time - under 65ms). Big Apple, anywhere within 20 miles - under 10ms.
Again, the above stats were round-trip - you don't need to multiple them by two  )
We're NOT co-located with any providers - we operate our own data-center (with all underlying infrastructure, under 24x7 video surveillance) with redundant air-conditioning and AC power-plants; our dual-redundant AC sources reside on the same AC power grid with NY stock exchange (which is just across Broadway from us). In fact, we provide co-location, IP and TDM connection as well as origination and termination services to many international operators. |
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ptrowski
Got Helix?
Premium
join:2005-03-14
Putnam, CT
clubs: | reply to jugg
If I am reading it correctly, Voipo allows you to toggle either straight to the gateways or allow them to handle the seesion. |
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VOIPoTim
VOIPo.com
Premium,VIP
join:2006-06-06
Houston, TX
1 edit | said by ptrowski :If I am reading it correctly, Voipo allows you to toggle either straight to the gateways or allow them to handle the seesion. Right.
By default, we don't proxy any audio unless there is a specific need to (certain NAT types).
Users can toggle audio handling in in their vPanel to force proxying through us. If set to be handled by us, it would be proxied through the VOIPo node the user is registered to (Dallas, Los Angeles, or Washington DC). The node is also selectable under "Network Selection".
For most users, having it go direct will result in the best audio, but it is dependent on the user internet connection, so there are some cases where users have great connections to us and it's more consistent for them. |
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