Betamax Servers for P-src-ip

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Aaron
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Betamax Servers for P-src-ip

Post by Aaron » Tue Dec 11, 2007 1:17 am

The list of Betamax servers that are getting the P-src-ip header set are:

voipstunt
voipcheap
voipbuster (will also match voipbusterpro)
voipdiscount
lowratevoip
webcalldirect
voipraider
justvoip
12voip
nonoh
poivy

For additions please request on this thread.

P.S. I don't think Betamax will allow this mechanism to last forever so don't get too attached...
Last edited by Aaron on Mon Dec 17, 2007 12:17 am, edited 3 times in total.

satphoneguy
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Post by satphoneguy » Tue Dec 11, 2007 2:09 am

please add:

webcalldirect
VOIPraider

i get the distinct impression that betamax is moving away from freedays on SIP enabled services. this is somewhat consistent with the industry as a whole two years ago nearly everything worked with(even if is was not advertised.) increasingly providers are offering separate plan for sip termination and residential plans including some free calling. the plans including free calling increasingly do not support SIP while the pay as you go do.

if the bulk of free minutes end up getting used via PBX's in call centers I can see why this is happening.

Aaron
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Post by Aaron » Tue Dec 11, 2007 6:25 am

satphoneguy wrote:increasingly providers are offering separate plan for sip termination and residential plans including some free calling. the plans including free calling increasingly do not support SIP while the pay as you go do.
Hi satphoneguy,

By providers do you mean Betamax offerings? If so I wouldn't be suprised if you are correct, still beats me how they are making money. Even Skype can't give away free calls.

As far as the industry in general goes I don't see SIP doing anything but getting more support. These days you can get use SIP to interconnect with ever a Tier 1 Telco down to a single end user ATA. If anything the momentum behind SIP is growing and hopefully one of these days Skype will open up their network for SIP traffic as well.

Regards,

Aaron

satphoneguy
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Post by satphoneguy » Tue Dec 11, 2007 6:45 am

i was speaking of all providers in general. i see sip defiantly increasing in support. but i am not convinced it is the 'platform for the masses' until setting stun servers and dealing with NAT issues and incompatible codecs, etc. are not issue it will be for the hobbyist(of whom i am one) and the enterprise customer with on sight technical support or using a hosted platform. i actually think the google jingle platform could gain momentum especially in peer to peer where there is no pstn link for the simple fact that it is more clearly defined. we could end up with two major standards one that involves the pstn and billing mechanisms(SIP) and another strictly peer to peer(jingle.)

as far as skype the reason I do not see them opening up to SIP or another standard is looking at the direction they are taking with 3 skypephone. in particular to get support from mobile operators around the world they would be making the right move to not support skype in/out in that setting and this would be hard to do with an open standard. if it is simply an alternate calling system that does not interact with regular phone numbers it suddenly becomes an interesting product to bundle into the cell phone package. on the other hand the operators would be crazy in places like europe where they earn so much from interconnection fees to suddenly allow everyone to receive calls on skypein. once SIP is brought into the picture it becomes way too easy for a third party to provide cheap land line numbers and outbound termination that directly take away from the operators income. we will find ways to use SIP on our mobile phones anyways; but the operators and not going to help that along. skype wants to be the friend of the operators and work along side them not something like fring or truphone.

satphoneguy
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Joined: Tue Oct 23, 2007 12:54 pm

one more reason why skype will not adopt sip

Post by satphoneguy » Tue Dec 11, 2007 7:02 am

one more thought to add to the above post:

more than anything else skype is a softphone/client application. they have nowhere near the best rates(at least the pay as you go rates;skypepro is not a bad deal) for termination. overall sip users prefer to use hardware ATA or IP phones. skype adopts sip and half the users switch to ip phone or ata's and they have suddenly become a proxy server and users will have all the issues with firewalls etc. that we sip users have(and they are not ready for it; although i rarely use it myself i recommend skype or gtalk to all my less technically inclined friends and family)

what would make sense is for skype to allow outbound(inbound is a more complicated issue since you could than point DID numbers) calls to sip uri. if i am not mistaken all skypein and skypeout calls already use sip under the hood. but this could be done without allowing registration of devices, etc.

Aaron
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Post by Aaron » Tue Dec 11, 2007 12:39 pm

satphoneguy wrote:i was speaking of all providers in general. i see sip defiantly increasing in support. but i am not convinced it is the 'platform for the masses' until setting stun servers and dealing with NAT issues and incompatible codecs, etc. are not issu ...
I'd agree with that to an extent. NAT is certainly a big hindrance with SIP although in controlled environments it can normally be coped with pretty well. For example we don't require STUN servers or any special settings to handle NAT for Blueface customers. But once that controlled environment starts to break down a bit for instance where an ISP chooses to use a transparent NAT or where a customer is using multiple routers and then all of a sudden there are two NAT'tings to cope with it get can hairy.
satphoneguy wrote:as far as skype the reason I do not see them opening up to SIP or another standard is looking at the direction they are taking with 3 skypephone...
<snip>
...operators and work along side them not something like fring or truphone.
I don't think mobile operators are that scared of VoIP. They stand to generate more revenue from their data plans than they do from international calls. I don't know the exact figures but I'd be suprised if international call revenue on a mobile network was more than 20% of their total. Again in the Blueface capacity we've dealt with a few mobile companies exploring VoIP (SIP in this case) and they have definitely not been scared of it and are trying to see where it fits into their offering in the same way ISP's do.

The Skype move is a good one by 3 for marketing, the technical side is not that significant. Skype on a mobile network won't be able to leverage the peer-to-peer aspect it can on PCs and it's also CPU intensive so it's actually not as suited technically to a mobile model as a SIP VoIP model is.

IPv6 will make a big difference for VoIP (and a lot of other applications especially real-time ones). It will allow NATs to be removed and the internet to operate as it was originally intended. Personally I can't wait for that day!

Regards,

Aaron

Billy the Bear
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Post by Billy the Bear » Tue Dec 11, 2007 6:20 pm

Dear Aaron,
Could you please add to the list:
voipbusterpro (www.voipbusterpro.com)
Thanks!

btw, guess you already know, at http://www.ip-phone-forum.de/showthread.php?t=86424 a 'complete' list of Betamax providers can be found (in German).

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TheFug
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Location: The Netherlands, North-Holland

Post by TheFug » Tue Dec 11, 2007 6:30 pm

Aaron, i've made an translation of your first post in this thread, in the Dutch forum section....
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

Aaron
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Post by Aaron » Wed Dec 12, 2007 12:18 am

TheFug wrote:Aaron, i've made an translation of your first post in this thread, in the Dutch forum section....
Thanks a lot.

Regards,

Aaron

hkr
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Post by hkr » Sun Dec 16, 2007 5:56 pm

Might be interesting with JUSTVOIP as well. Although they do not allow SIP for outgoing, they do allow SIP incoming, at least for certain IPs (e.g. Voxbone) . I do not know if Voxalot is on their list of accepted IPs or not...

By the way: has anybody experimented with the Betamax call forwarding capabilities? If I divert my DID to Voipdiscount, justvoip or smsdiscount, they will work fine in the sofphone or SIP phone registered with my Betamax accounts. However, if I set call forward from these betamax services, incoming SIP calls will indeed ring the destination (fwd) number, connection is established, but there will be no sound... I recently purchased a JustVoip DID. If I get a call to that number by PSTN, it will forward properly, but if call arrives to Betamax by SIP, there is no sound.

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