Draft:Traditional phone to ToIP gateways
Submission declined on 28 September 2024 by KylieTastic (talk). This submission reads more like an essay than an encyclopedia article. Submissions should summarise information in secondary, reliable sources and not contain opinions or original research. Please write about the topic from a neutral point of view in an encyclopedic manner.
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- Comment: Reads more like an essay and original research. It covers a few specific cases and misses lots about PSTN to VoIP gateways in general. Several parts are about VoIP to VoIP and encyclopaedic articles don't have Tricks and Observations. KylieTastic (talk) 14:57, 28 September 2024 (UTC)
Traditional phone to ToIP gateways are a sort of communication services, whom permits to a classic phone subscriber (landline, cellular or satellite) to communicate at a local price, to a Voice over IP (or Telephony over IP) user, whom doesn't have a classic phone number.
However with the democratisation of the internet access, and especially due to the large marketshare of communication apps, in the 2000s on computer, and years later on mobile devices, those solutions are generally unknown from large public.
History
[edit]This type of system benefits of the arrival of the Telephony Over IP protocols, such as SIP for VoIP, or Jingle for XMPP protocol. It permits to let internet users to communicate for free and unlimited, generally through independent and low-cost phone providers, whereas outgoing calls to PSTN is charged.
It's a bit the same of Skype, where only from internet to internet calls are free of charge. However, calls to the PSTN are always charged, but at a different price. And Skype or others Whatsapp (lately) requires the both correspondent to use the same software, where SIP or XMPP users can choose different client-software, and even communicate with others providers's subscribers.
It's the older one still in service. Started in 2005, managed by an Australian consultant, it still process about hundreds of calls every year. This service is financially maintained by various providers of Voice over IP, whom are not big phone providers with consequent infrastructures, but small or medium business retailing time of call. Their customers are mainly individuals, small and medium organisations.[1]. They have a certain success to a small range of customers, but bring to them for free to SIPBroker, several gateways, free to call, all over the world. Those small providers are mainly DIDWW, voip.ms, callcentric and several others[2].
Functionning
[edit]A PSTN to SIP gateway, is a computing-server, accessible from the common phone network. A public landline phone number has been attributed to it. For example, +34-97-157-99-12, located in Spain, allows to reach a VoIP user located everywhere in the world. However some attention has to be ported to the latency[3].
When the communication to the gateway, using a traditional phone number, is established, the caller must compose the asterisk (or star, *) on the keyboard, then the network (voip provider) number, then the voip user number[4]. A VoIP number, is also called alias or DID[5].
Example, to illustrate :
[edit]Alice, get a VoIP number, 9839, from her voip.ms provider, which is the 228 network number. Thus, when the communication will start the robot's voice, bob will have to enter *2289839 on it's phone's keyboard. Until processed in few seconds by the gateway, if Alice's phone is connected, her phone will ring.
The network number, depending of the VoIP provider, could be obtained on the SIPBroker website[6].
Steps :
- Firstly, the phone number of the most adapted gateway (e.g. the country) might be composed on the phone. Those gateways can be found on the sipbroker website[7].
- When the robot starts its voice, enter the star/asterisk * , then the VoIP service's network number. It can be found too, on the sipbroker website[8].
- If the VoIP's user is online, the phone will ring.
In addition of being reachable via the PSTN-to-SIP gateways, Alice, at a specific VoIP provider (e.g. voip.ms), can be reached from others providers, such as callcentric, DIDWW, etc., for free and unlimited, e.g. like skype, but using any type of SIP-client : so alice ('at\) first . org could communicate unlimited and for free with bob ("at\) last . net, because SIP protocol is interoperable[9].
Advantages :
[edit]- Those communications are generally unlimited, and toll-free by VoIP brokers[10]
- Those pstn-to-sip gateways are present in lot of countries (about half of the world, where North America have a lot)
- It doesnt requires the caller to have a smartphone or even a mobile phone, it will even work on a landline.
Disavantages :
[edit]- Like any call on the PSTN (traditional phone), it can be tapped.
- Some gateways have a maximum capacity of simultaneous calls.
- Less than half of countries in the world have a gateway ; In Asia, only Japan have. But generally countries all over the world have providers that permits free calls to border countries.
- Some rare providers in the world, toll (means apply fees) the calls to this type of services. This is not due to the VoIP providers, but because their commercial rates applies fees to any call not going to mobiles or landlines.
- This type of service could be considered as obsolete, mainly due to the mass adoption of smartphones in the world. Also, some countries doesn't charge fees for calling to borders or others countrie's landlines calls. However, not all providers, all over the world, accept incoming calls to landlines for free, eg Switzerland mainly applies fees to calls to any number, even landlines.
At the end of the 2010s, the massive adoption of mobile apps, especially Whatsapp , is a bit shaked by contracts modifications[11]. In result of this, millions of users decides either to leave or delete Whatsapp , or to add another one, or just move to another service[12]. Some others types of communications, started few years ago, starts to gather amount of new users, such as Matrix protocol, but also XMPP. This last one, created at the end of the 90s, supports VoIP since 2005, and is now ready to compete for bigger apps. For example, Monal on iOS, Conversations or Quicksy for Android/AOSP, Dino for Windows/Linux/BSD/Macos, whom supports Voice over IP and Videoconference since years. A new type of operator based on XMPP is started : JMP.chat.
JMP[13] permits to register and redirect a North-American phone number (+1 prefix) to a specific xmpp address[14]. The subscription is cheap, and benefits of the popularity of smartphones, however it's possible to connect to JMP.chat from any xmpp-compatible device (computer, tablet..)[15]. This service is provided by the non profits Canadian association of Soprani[16], to promote only-internet based communications with interoperability and standards used. In addition, JMB permits its subscribers, and xmpp users in general, to be reached via the SIPBroker network, but the process is not convenient ; the xmpp address have to be converted in numbers-only chain, whom can take too much time to be typed at the robot's voice, resulting in timeout effect[17].
At the end of the 2000s, several actors started the iNum standard, in case a virtual phone number starting with +882 and +883 prefix[18], to permit to internet users everywhere in the world to be called for free[19]. But even with that initiative, firstly the traditional network operators where not playing fair game, charging those communications[20] (from traditional (pstn) to iNum users) at a higher price, such as satellite communication, to discourage this practice. The massive adoption of internet-mobile application, such as Whatsapp, Signal, Telegram, iMessage and RCS commercial-based alternative, definitely stopped the maintain of the service[21]. Thus, the iNum service was shuted down a decade later[22].
That service was provided by Voxbone, available through few retailers such as Ippi, whom provided free iNum numbers to it's users, even without paying, and was standardised by ITU[23].
Tricks :
[edit]Uses for non-volte compatible devices
[edit]A trick not well known consists to use the PSTN to SIP gateways to bypass the lack of software-support of Voice over LTE for non-compatible devices. The trick consists to use the alias provided by the VoIP service, to register it into the phone through SIP-software client app. Thus, calling from PSTN to the VoIP number will make Voice over LTE possible. However, this system is totally independent of the cellular service : it will not magically enable VoLTE, but enable a different number, permitting from communucations to VoIP users through gateways, to be reached from common phone network. It's mainly used in case of absence of 2G coverage, 3G coverage, or just to permit calls on non-VoLTE devices still using 4G service.
Uses as VoWifi device
[edit]The same trick could be used as a sort of VoWifi call, requiring to be connected and close of a Wifi AP. It permits to reuse non-volte or non-vowifi compatible phones, as a device to receive calls, or do directly through the internet.
Observations
[edit]Some engineers let think that both SIP and XMPP protocols, are for messaging, voice and video over IP, what email is for, especially because they work on the same scheme.
It's interoperable, decentralized, as it's possible to send email from "first . org" domain to "second . net" one, mean between different service providers, contrary to popular messaging apps, whom requires having the app for each participant, and connected to a unique service provider.[24]
It's the same functionning for XMPP and SIP , as messaging, voice and video communications. The only case is in case a service provider isn't compatible with a feature, which is rare. Mainly, they all accept voice, video features, and file sharing or encryption features. In addition, those services are "domain-opened", means that providers's users can communicate each-other, but it appears some VoIP operators are still domain-closed for specific reasons, or bills all calls, even to direct internet users, but it's rare[25]. In addition, even if VoLTE and VoWifi works with SIP protocol to carry Telephony, they will never accept being called from direct VoIP communication, requiring incoming calls coming from traditional phone network, not internet users.
References
[edit]- ^ Company, Australian Phone. "VoIP (Voice over IP provider) | Australian Phone Company – Business VoIP Providers Australia | Cloud / Virtual / Hosted PBX | SIP Trunking | Residential VoIP | Best Australian VoIP Provider | australianphone.com.au". www.australianphone.com.au. Retrieved 2024-07-26.
{{cite web}}
:|last=
has generic name (help) - ^ "Wacky Tel". wackytel.appspot.com. Retrieved 2024-07-26.
- ^ Info, VoIP (2005-12-30). "SipBroker". VoIP-Info. Retrieved 2024-07-26.
- ^ "Australian First Mobile Phone Activated VoIP Telephony Service - VoIP Monitor". www.voipmonitor.net. Retrieved 2024-07-26.
- ^ "https://www.callcentric.com/faq/49".
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: External link in
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- ^ "https://wiki.voip.ms/article/FAQ#What_is_a_DID.3F".
{{cite web}}
: External link in
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- ^ "SIPBroker - PSTN Numbers". sipbroker.com. Retrieved 2024-07-26.
- ^ "SIPBroker - Provider White Pages". sipbroker.com. Retrieved 2024-07-26.
- ^ "Introduction to SIP – www.pasta.freemyip.com". www.pasta.freemyip.com. Retrieved 2024-07-26.
- ^ "https://www.ippi.com/".
{{cite web}}
: External link in
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- ^ Hern, Alex (2021-01-24). "WhatsApp loses millions of users after terms update". The Guardian. ISSN 0261-3077. Retrieved 2024-07-26.
- ^ "WhatsApp exodus sees new Telegram users rise 500%". The Independent. 2021-01-13. Retrieved 2024-07-26.
- ^ "JMP.chat". jmp.chat. Retrieved 2024-07-26.
- ^ "ThePlan - Soprani.ca Wiki". wiki.soprani.ca. Retrieved 2024-07-26.
- ^ "JMP: Frequently Asked Questions". jmp.chat. Retrieved 2024-07-26.
- ^ "https://soprani.ca/".
{{cite web}}
: External link in
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- ^ "Cheogram SIP". sip.cheogram.com. Retrieved 2024-07-27.
- ^ "http://www.ietf.org/rfc/rfc3261.txt".
{{cite web}}
: External link in
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- ^ "Blog | inum | inum – One number for the world". 2018-12-04. Archived from the original on 2018-12-04. Retrieved 2024-07-27.
- ^ "https://www.voxbone.com/blog/future-of-inum/".
{{cite web}}
: External link in
(help)|title=
- ^ "Localphone - No More Free iNums - VOIP Tech Chat | DSLReports Forums". DSL Reports. Retrieved 2024-07-27.
- ^ "Voxbone decommissions iNum - VOIP Tech Chat | DSLReports Forums". DSL Reports. Retrieved 2024-07-27.
- ^ "rise-inum-international-number".
- ^ "slides-84-dispatch-1" (PDF).
- ^ "Introduction to SIP – www.pasta.freemyip.com". www.pasta.freemyip.com. Retrieved 2024-07-27.