sip2sip.info seems to be great service for any testing purposes and also nice free VoIP operator in general. I remember trying few free (totally free) VoIP services about 10 years ago, but I think none of them worked well in typical scenario with client behind the NAT and/or with misconfigured public IP address. They were not handling media streams at all, working just as SIP proxy/registration server, which of course was understandable given they were operating with no cost for end user. As sip2sip.info handles NAT, it just works without any requirements for the client configuration.
From my short testing:
- creating account is painless and quick
- even without account you can call sip:firstname.lastname@example.org and sip:email@example.com - just dial it with tSIP, out of the box
- even without account you can dial to any any ad-hoc conference room (call sip:firstname.lastname@example.org)
- if you have account and softphone registered anyone can call you, even if does not have account at this operator
- Opus works, Speex works
- call parties much have matching codecs selected
- text messages are working and can be sent to sip2sip.info even without registering account, but
- if sending message to endpoint that is not registered at the momemt I'm seeing different status (202 instead of 200) which should indicate that message is waiting to be relayed, but when receiving softphone is registered later it is not receiving message
- messaging to Blink is not working
- sending message from Blink creates call with m=message media line - tSIP does not support MSRP
In general: two sides of the call must be interoperable - using common codec(s) and (if needed) same way of sending text messages (either SIP SIMPLE or MSRP).
iptel.org seems to be another free and easy to use VoIP operator. For test call sip:email@example.com can be used (works even without creating/using account from this operator).
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