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Author Topic: SIP-only providers: compare OnSIP, SIP2SIP, ippi  (Read 1978 times)
SeanTek
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Posts: 32


« on: August 11, 2017, 02:02:25 am »

I am looking for a SIP-only provider (no PSTN interconnection needed) for my Obi1062 and Obi200, and am trying to compare between OnSIP, SIP2SIP, and ippi. (PSTN interconnection is already handled by GV and Callcentric.) I would like to try to get more reliable Opus HD calling and (perhaps) encryption via SRTP/ZRTP. Based in USA so slight preference for US-based services.

What are people's experiences with these providers, now that it's 2017? I would rather set up just one account, instead of having one on each service and have to remember or juggle. I have a GetOnSIP account but I understand that GetOnSIP is deprecated so if I want to use the "latest" features I should migrate to OnSIP Free.
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drgeoff
Hero Member & Beta Tester
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Posts: 2485


« Reply #1 on: August 11, 2017, 05:20:14 am »

What exactly are you wanting to achieve?

I have several 'SIP only, no PSTN in or out' accounts but they never get used for real calls.

If you want to call to and/or be called by SIP endpoints you can probably do that directly on the OBis without needing a SIP only provider. But a free account at PBXes or the like may make some of the network security issues easier.

Using SIP end to end without any providers in the route means that choice of codec and encryption is not subject to their constraints.
« Last Edit: August 11, 2017, 05:27:13 am by drgeoff » Logged
SeanTek
Jr. Member
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Posts: 32


« Reply #2 on: August 11, 2017, 09:50:31 am »

What exactly are you wanting to achieve?
[...]
Using SIP end to end without any providers in the route means that choice of codec and encryption is not subject to their constraints.

Yes, that: Choice of codec and encryption are not subject to their constraints (or specifically, the PSTN's constraints). A nice bonus would be to use my own domain name with SRV records so that someone could (theoretically) call me using my e-mail-like addresses. I believe all three named services support that option.

I would have just gone with OnSIP, but their WebRTC client seems to think that all-numeric identifiers are PSTN phone numbers, even though @domain.com is placed afterwards. This means that Callcentric and iNum SIP addresses don't work. I put in a support request. I suppose that this is not a limitation of OnSIP in general, but it caused me enough pain to start to look around for other options. ippi touts Skype and Google Hangouts integration (not sure how reliable). SIP2SIP hits all the right SIP-only notes, but I am less sure of reliability and longevity.
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drgeoff
Hero Member & Beta Tester
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Posts: 2485


« Reply #3 on: August 11, 2017, 10:05:39 am »

You are still reliant on the other end having, and agreeing to use, the codec and encryption. And you don't get the full benefit of any potential better sound unless both ends have audio transducers and circuitry designed to exploit it.

I've played with ippi's Skype gateway a few times in the past and found it not always 100% working.
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SeanTek
Jr. Member
**
Posts: 32


« Reply #4 on: August 11, 2017, 10:12:00 am »

Right. So basically...other OBi 1000 devices. Tongue Maybe Grandstream if we're lucky.

Ultimately, both parties have to be using IP phone equipment or softphones, and be telecom nerds. Or they can just use FaceTime...or Skype...or WhatsApp...or Ooma...or Google Hangouts...or...(basically any closed system where both parties are using the same technology at the endpoints).
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