[HN Gopher] 20 Years of SIP - A Retrospective
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20 Years of SIP - A Retrospective
Author : psim1
Score : 86 points
Date : 2022-06-09 18:50 UTC (1 days ago)
(HTM) web link (www.jdrosen.net)
(TXT) w3m dump (www.jdrosen.net)
| rr888 wrote:
| I'd love in SIP was more normalized for most people, so you just
| bought a data sim (and mostly used wifi) and didn't have to port
| a number around. I guess wireless market is so competitive the
| carriers really want to avoid this.
| jaywalk wrote:
| It's been well over a decade since I've ported my number, but
| even way back then it was a fairly simple and straightforward
| process. I doubt it's gotten more complex since then.
| julianlam wrote:
| At least in Canada, the process is simple but still not
| instantaneous, unless you are porting between the big
| incumbent carriers (Rogers, Bell, Telus, and their
| subsidiaries--often referred to as Robelus).
|
| I ported my number out to VoIP.ms, and it included a four day
| waiting period... for no particular reason except to probably
| allow for some human to check a box.
| phowat wrote:
| Absolutely, I remember owning a nokia n85 12 years ago which
| had native SIP support. I wish ios and android had this.
| Nextgrid wrote:
| Early Android had it - wouldn't be surprised if it still
| exists now but has been hidden from the UI. For VoWiFI/VoLTE
| it's already using SIP anyway so there definitely _is_ a SIP
| client.
|
| The only reason you're not given access to it is because
| carriers are doing their best to protect their obsolete
| business model, and despite the appearances, Apple is fully
| complicit as well.
| wolrah wrote:
| > Early Android had it - wouldn't be surprised if it still
| exists now but has been hidden from the UI.
|
| It's buried in the settings for the Phone app (the dialer)
| and AFAIK is often removed from OEM ROMs because of course
| it is.
|
| As a VoIP engineer, it's a terrible soft client. It works,
| barely, and has basically no features beyond bare minimum
| calling. I've tried to use it repeatedly over the years but
| always ended up on commercial softphones like Bria or GS
| Wave.
| kevincox wrote:
| SIP was actually removed from the ASOP dialer very
| recently. It was removed in Android 12 (released late
| 2021). The support was never great (it would reset to
| defaulting to SIM calls every reboot) but it worked and
| was reliable.
| bestham wrote:
| SIP is the basis for VoLTE and VoNR. So in some sense iOS
| and Android still support SIP.
| supertrope wrote:
| Although Google stopped supporting the OS integrated SIP
| client and eventually removed it, you can install Acrobits
| Groundwire or Bria. Those support PUSH notification for
| incoming calls. Push is better than missing calls because the
| app got killed, or forcing the app to run 24/7 and severely
| shortening battery runtime.
|
| But the call quality will never be as good as the native
| phone app as that gets QCI prioritization.
| supertrope wrote:
| In the US market average customers can barely handle SIMs.
| Letting customers handle their SIP credentials would amplify
| number hijacking and customer side telecom fraud.
| Nextgrid wrote:
| I don't see any cases where customers are misusing SIMs that
| would expose them to fraud at a large-scale. What I see
| instead is customer service being staffed by monkeys that are
| too stupid to realize they're being social-engineered, don't
| care or are outright complicit in the fraud.
| supertrope wrote:
| Customer side as in whoever controls the
| SIM/credentials/handset. It would also need an inordinate
| amount of support. Customers are used to SIMs being
| preinstalled. Just imagine customers forgetting their SIP
| password or being phished for them. Or using Password1!
| ggping wrote:
| Having worked in customer service decades ago, I wouldn't
| call them stupid monkeys. Designing your 2F authentication
| around SIM-based telcom is perhaps more appropriate to
| obtain the title of "stupid".
| Nextgrid wrote:
| I'm sure there are good ones in the mix but that doesn't
| match my experience with any mainstream consumer-grade
| carrier or ISP. It doesn't _have_ to be that way but
| obviously until carriers are held accountable and /or
| their oligopoly is broken they have no incentive to
| improve things.
|
| While it's true that SMS 2FA is flawed, that still isn't
| an excuse for letting customers' phone numbers being
| taken over by very unsophisticated attacks, sometimes
| even if notes are added to the account (or a PIN) that
| explicitly warn against such attacks.
|
| Also, I'm not sure how much of "decades ago" is hyperbole
| but back in the day ISP/telco support was a great career
| path and would allow you to learn and move up the ranks
| towards a more technical position. Nowadays "support" in
| any customer-grade ISP/telco is a dead-end position
| that's there to be exploited as much as possible (in fact
| it's often outsourced to a boiler room abroad, probably
| right next to the tech-support scammers) and replaced by
| a new sucker as soon as you burn out. Obviously this kind
| of treatment doesn't attract the right talent nor inspire
| goodwill in said talent.
| achillean wrote:
| The number of publicly-accessible SIP services has decreased
| significantly over the past few years:
|
| https://imgur.com/a/tqXclNi
| zajio1am wrote:
| One sad thing about SIP that despite being pretty common here in
| Czechia, it is only used as a last hop to PSTN instead of as an
| independent federated network.
|
| It makes economic sense - it is hard to monetize running SIP
| servers for independent network (and one cannot use ads like with
| e-mail as SIP clients are not web apps), but you can monetize
| selling access to PSTN.
|
| Today, with WebRTC, one can build web client for SIP, but WebRTC
| VoIP services are still just silos.
| userbinator wrote:
| _and one cannot use ads like with e-mail as SIP clients are not
| web apps_
|
| Don't broadcast radio and TV have ads too, despite being
| independent of any client implementation?
|
| (I hate ads as much as anyone, but it's possible to run
| pre/inter-call ads on a free call too.)
| userbinator wrote:
| On the other hand, there are now many proprietary voice/video
| communications platforms NOT based on SIP, with some of them
| adding SIP support only as an afterthought/additional-cost
| feature. I don't think the authors of SIP envisioned this, and
| it's unfortunate. It was intended to be as widespread as email,
| with a similar diversity of implementations of varying
| interoperability.
| Sean-Der wrote:
| It is a shame. I don't see any easy answer. I am working on
| sfu-to-sfu[0] and hope it can make some traction. If we can get
| all the Open Source WebRTC servers working together, maybe
| there is hope? I believe WebRTC did the right thing. It was as
| flexible as possible to make it more palatable. You can
| standardized/codify things, but you can't undo it :)
|
| I am also really excited about WHIP[1]
|
| [0] https://github.com/matrix-org/sfu-to-sfu
|
| [1] https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
| psim1 wrote:
| It would have required much forethought to predict, in 2002,
| that realtime communications would be happening in the browser.
| Thus WebRTC was born. But a tragedy of WebRTC is that it is not
| by-default compatible with SIP. WebRTC authors could have
| specified SIP as the signaling protocol over websocket, but
| they left the spec open ended. I feel two ways about this: glad
| for choice, but disappointed that the choice most often made
| does not allow for interop with the "legacy" realtime
| communications protocol, SIP.
| kkielhofner wrote:
| There are libraries to do SIP signaling over WS/WSS with
| WebRTC from a browser (or whatever) that theoretically allow
| for interop with other SIP devices.
|
| In practice the fundamentals of WebRTC rely on things like
| ICE, STUN, and TURN for media so they're not going to be
| compatible with almost all existing SIP implementations -
| many of which can't even do interop with bog-standard vanilla
| SIP over UDP with standard codecs properly.
|
| As is often the case with SIP you're back to using some
| Session Border Controller or equivalent architectural
| component to make interop actually work reliably.
| csdvrx wrote:
| Can anyone recommend a good quality US-based SIP provider?
| (twilio is OK, but I need to connect physical Cisco phones)
|
| I'd like to port my phone numbers from google, as I'm afraid the
| migration of the free domain may cost me my phone number in case
| of shenanigans (like google voice being considered separate of
| google mail etc)
| anderiv wrote:
| 10-year business customer of https://voip.ms/ here. I have
| nothing but great things to say about their service itself, its
| reliability, and their support. It's also very inexpensive for
| what you get.
| toast0 wrote:
| Technically, voip.ms is CA based, not US based. But they've
| got a lot of US infrastructure.
| F00Fbug wrote:
| Callcentric provides a bunch of features for a great price. I
| ported over my two landlines over 10 years ago and love it.
| When my mom moved to a new state, I ported her number first so
| she could keep that number in her new place.
|
| If you're actively managing a large number of users and
| devices, I had great luck with OnSIP. They're not the cheapest
| game in town, but their management interface is top notch. They
| were always innovating and the architecture they disclosed was
| impressive; very focused on HA and performance.
| mandrill wrote:
| twilio has a few competitors--telnyx, bandwidth, plivo, vonage
| just name a few that you can port a number to for SIP.
| csharpminor wrote:
| FYI Twilio has a few guides on how to connect their SIP
| trunking product to Cisco SBCs:
|
| CUCM: https://twilio-cms-
| prod.s3.amazonaws.com/documents/InteropGu...
|
| ISR: https://www.twilio.com/docs/sip-trunking/sample-
| configuratio...
|
| They also have a porting process you can use to migrate your
| numbers from GV/Bandwidth (or you can just buy a Twilio number
| for $1).
| leitess wrote:
| js2 wrote:
| I've been happy with Anveo for years now.
| michael_michael wrote:
| I hate to be that guy, but if you don't mind being a Linux
| sysadmin you might consider running your own Asterisk server. I
| do that for my SMB with about a dozen physical sets, and use
| Twilio for SIP trunking.
|
| I followed the first 10 chapters or so of the O'Reilly Asterisk
| book making a few changes here and there to suit my preferences
| (different Linux flavor, different DB). I run a $10/month
| Digital Ocean droplet that hosts the Asterisk server. If you
| can deal with config files, you can have a rock-solid PBX with
| Enterprise-grade features for the cost of the server + Twilio's
| SIP trunking features. It ends up costing about $25 every 1.5
| months or so. I barely ever think about it, except when I need
| to tweak a greeting for holiday hours or something.
| Nextgrid wrote:
| Asterisk is a huge pile of C code with lots of legacy crap. I
| wouldn't trust having it exposed to the Internet.
| ninjin wrote:
| Happy SIP user for nearly twenty years, which allows me to bridge
| three countries. Currently using baresip [1] and finding it to be
| remarkably reliable, but is there any hardware phone out there
| that I can put on my desk? Or is the sane thing to do to get a
| handset and hook it up to a computer via say USB? I have tried at
| least twice over the years to gain some clarity on these
| questions, but maybe I am using the wrong search terms?
|
| [1]: https://github.com/baresip/baresip
| js2 wrote:
| If I understand what you're asking, I use an Obihai VoIP
| adapter so I can use any old phone, but there are also a
| variety of IP phones from Cisco, Obihai, etc.
| Nextgrid wrote:
| Most business-grade desk phones support SIP.
| anderiv wrote:
| This is true. I will give the caveat, though, that some of
| these are vendor-locked. Meaning: you can't easily use them
| with 3rd party SIP providers.
|
| So, just be aware of this and do your homework on specific
| brands/models before purchasing to ensure you'll get
| something that will work for you.
| anderiv wrote:
| There are _tons_ of SIP-compatible phones out there. If all you
| need is access to a single SIP account, the Grandstream GXP1610
| is very inexpensive (~$40 US) and will do the trick. They also
| have more expensive models that support SIP accounts.
| F00Fbug wrote:
| Polycom phones are really great... I deployed VOIP for my
| employer some years ago and put in about 40 Polycom devices in
| 4 states. They're not cheap, but full featured and very well
| made.
|
| You could also get an ATA (https://www.amazon.com/Grandstream-
| HT801-Single-Port-Telepho...) and plug a traditional phone into
| it. I used one of these at home for a long time. Just realized
| it's still plugged in an running and I threw out my last analog
| phone over a year ago!!!
| Terry_Roll wrote:
| And ATA's are an excellent backdoor into computer networks
| because the caller ID uses an old dialup modem protocol...
| Nextgrid wrote:
| Could you elaborate? What's the attack vector here?
| ninjin wrote:
| A big thank you to everyone responding with information,
| apologies for responding only here. It looks like there is
| indeed still a lot for me to learn, but now I have some
| pointers. I have been meaning to get my hands dirty with
| SIP for some time, dreaming of a setup with multiple
| accounts and control over things like when each account
| allows incoming calls, etc. But, as Terry_Roll indicated,
| there seems to be plenty of security considerations as well
| which makes me somewhat uncomfortable.
|
| Also found the /r/VOIP subreddit [1] which has plenty of
| reading.
|
| [1]: https://teddit.net/r/VOIP
| Ekaros wrote:
| I think issue with SIP is that it is and was driven by the
| industry. And I wouldn't call it exactly simple or easy protocol
| to follow and implemented. Thus many alternatives or self-made
| solutions are more likely to be chosen. It does a lot, but at the
| same time it is increasingly complex protocol. At least compared
| to others.
| kkielhofner wrote:
| To me the fundamental issue with SIP is the recognition by
| vendors that a standard protocol is "here" while still being
| incentivized (of course) to implement vendor lock-in. It was
| also clear that SIP for trunking (with very basic call
| setup/teardown) was the path forward to replace PRI, POTS, etc.
|
| Consider government, large corp, etc purchase requirements. "Oh
| SIP is the standard. Cisco do you support SIP?" Cisco says "Of
| course!". Check the box and buy.
|
| Meanwhile at the time their ecosystem is 99% Skinny (their
| proprietary protocol) and SIP is an afterthought for anything
| other than extremely basic call functionality, trunking, etc.
| Even when pushed to release SIP firmware for their (at the
| time) $500 hardware phones the SIP firmware was so feature
| crippled you're literally throwing money away by using it.
|
| So everyone installs Call Manager to be done with it and have
| something that actually works. Even when Cisco got around to
| essentially being SIP native getting 30 year old features like
| hold, transfer, busy lamp fields, provisioning, etc working
| between vendors was nearly impossible.
|
| Repeat for just about every implementation in existence.
| psim1 wrote:
| Why is industry-driven an issue? And compared to what others?
| On the contrary, SIP is as readable and understandable as HTTP.
| Someone who understands HTTP can learn SIP-based VoIP quite
| easily using their software/web background without needing a
| telecoms background.
| jimmySixDOF wrote:
| To the extent SIP was industry driven, it was due to the fuzzy
| standards themselves which left a lot of room for vendor
| specific handling especially used to max out their own device
| to device performance often supported by proprietary
| enhancements. The biggest trade show was called Interop for a
| reason. At this point SIP might have served its purpose, but
| the basic approach of separating a control signaling plane from
| media paths has stood the test of time.
| julianlam wrote:
| SIP has been an amazing standard that untethered me from the
| triopoly of cellular providers in Canada.
|
| For the cost of a data-only SIM ($15/mo), I can call, text, and
| surf. I only need to be wary of the 3gb cap.
|
| For those in EU/Asia, can you believe that here, that is
| considered an amazing deal? It's still unfathomable outside of
| North America, but imagine that everybody else pays at least 4-5x
| more than I do.
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