PoloDude Premium Member join:2006-03-29 Aiken, SC |
to HeadSpinning
Re: [Northeast] ISDN BRI possible over FiOS?HeadSpinning - I can assure you that nycdave knows what he is talking about. |
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said by PoloDude:HeadSpinning - I can assure you that nycdave knows what he is talking about. I'm sure he thinks voice riding on a fibre optic cable is analogue, but it is not. His statement was that FiOS is analogue voice. It is clearly not. If it were analog voice, it would not require a digital adapter at the home nor would it require a battery supply at the home. It is voice derived from a digital connection to the home. That is fundamentally different from an analogue tip and ring connection served from a central office or SLC in that it requires local terminating equipment and power at the prem and has the associated limitations. |
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nycdave MVM join:1999-11-16 Melville, NY |
Guess what? The FiOS ONT is just like a SLC for voice....It is an updated SLC or RT. So I guess by your definition of a SLC that FiOS voice is analog..... |
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said by nycdave:Guess what? The FiOS ONT is just like a SLC for voice....It is an updated SLC or RT. So I guess by your definition of a SLC that FiOS voice is analog..... You also said that SIP and FDV are analogue when clearly SIP has an IP address and uses VoIP... "All FiOS voice is analog voice - no VoIP. SIP and FDV are still considered analog voice, since the ONT provides analog dial tone...." Just because the Verizon marketing people want to position FiOS voice as analogue doesn't make it true. While I agree that FiOS digital voice clearly is not the same animal as Magic Jack and Vonage, and provides as close an emulation as possible of an analogue service as can be provided on a packet or cell switched network, if it is analogue because it provides a jack at the house to plug in a POTS phone then by THAT definition, so are the rest. ATM BLES is exactly that - Broadband Loop EMULATION, and SIP ... If you're debating that SIP is not IP, then you really need to do some reading. |
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nycdave MVM join:1999-11-16 Melville, NY |
Well I'm not here to debate SIP is IP when it clearly isn't....SIP and IP are 2 different protocols.
SIP employs design elements similar to the HTTP request/response transaction model.[5] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.
From Wikipedia. |
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mackey Premium Member join:2007-08-20 |
to HeadSpinning
said by HeadSpinning:I'm sure he thinks voice riding on a fibre optic cable is analogue, but it is not. Um, it's quite easy to send an analog signal over fiber WITHOUT converting it to digital. The cable co's have been doing it for years. Just because it's fiber doesn't mean it's digital. said by HeadSpinning:If it were analog voice, it would not require a digital adapter at the home nor would it require a battery supply at the home. What? Analog electronics do not require power? And here I was changing batteries in my portable radios all those years! I hate to break it to you, but ALL electronics require power. Even your old analog phones require power. Even if the ONT didn't need it, you would still need a power supply to power all your phones as you cannot send power over fiber. /M |
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said by mackey:said by HeadSpinning:I'm sure he thinks voice riding on a fibre optic cable is analogue, but it is not. Um, it's quite easy to send an analog signal over fiber WITHOUT converting it to digital. The cable co's have been doing it for years. Just because it's fiber doesn't mean it's digital. The cablecos do use RFoG for analogue video, and also use QAM for digital video. That is true. But in the case of FiOS, the telephone and data is sent using digital technology. said by mackey:said by HeadSpinning:If it were analog voice, it would not require a digital adapter at the home nor would it require a battery supply at the home. What? Analog electronics do not require power? And here I was changing batteries in my portable radios all those years! I hate to break it to you, but ALL electronics require power. Even your old analog phones require power. Even if the ONT didn't need it, you would still need a power supply to power all your phones as you cannot send power over fiber. /M I was referring to local power. The obvious advantage to analogue phones is that the loop provides the power from the CO. FiOS is a digital telephone, not analogue. The signal coming in to the house is digital, it is converted to analogue at the subscriber's prem. Verizon simply wants to distance themselves from the OTHER digital services like Vonage and MajicJack that, although they employ similar technologies, have disadvantages. |
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HeadSpinning |
to nycdave
said by nycdave:Well I'm not here to debate SIP is IP when it clearly isn't....SIP and IP are 2 different protocols.
SIP employs design elements similar to the HTTP request/response transaction model.[5] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.
From Wikipedia. Correct, but SIP and HTTP use TCP/IP as their underlying transport mechanism. The media is sent over RTP streams which are transported on UDP/IP. SIP and HTTP run on the TCP/IP model - therefore they are IP. SIP is digital. I still find it amazing that you would take the position that "FiOS Digital Voice is Analogue". I suspect that despite the fact that you're an intelligent and knowledgeable person, you've heard others in your company repeat the mantra that "FiOS Digital Voice is Analogue Voice" and you never stopped to even think about how absurd that position is. |
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buckinghamDoylstown Pa Premium Member join:2005-07-17 Buckingham, PA |
to nycdave
said by nycdave:Well I'm not here to debate SIP is IP when it clearly isn't....SIP and IP are 2 different protocols. To expand upon Dave's comments, SIP (Session Inititated Protocol) is a signaling standard. It's transported over IP (Internet Protocol) The voice, video or other traffic that SIP provides signaling for can also flow over that same IP network, but the media is separate. ----- Relative to the original questions, BRI and PRI are TDM (time division multiplexing) type technologies with signaling on one channel and media over two or 23 channels, depending on whether or not it's a BRI or PRI. (basic rate interface or primary rate interface) While it's theoretically possible to transport them over fiber, (and many PRIs are), it's a whole different setup from FiOS. And I'm really surprised that the audio industry would still prefer to use BRI...the whole 128kb of bandwidth isn't likely very "special" for bandwidth intensive flows, especially in this age of much higher throughput services being available. I do believe it's used more around live broadcast of voice, but other content would really suffer. |
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to nycdave
NYCDave, First of all thanks for all the work you put in over the years here at dslreports. But I think we can end this debate on Fios Digital Voice quickly and clearly. From Verizon support page FiOS Digital Voice Service FiOS Digital Voice is a specific type of Voice Over Internet Protocol (VoIP). » www22.verizon.com/Suppor ··· 1150.htmIf Verizon is calling it voip why are you so admit that it not. It may not be sip it could be some other type of signaling protocol but it clearly not sending analog down the fiber the ont is switching it to packets and those packets making it to the co softswitch. |
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to buckingham
said by buckingham:Relative to the original questions, BRI and PRI are TDM (time division multiplexing) type technologies with signaling on one channel and media over two or 23 channels, depending on whether or not it's a BRI or PRI. (basic rate interface or primary rate interface) While it's theoretically possible to transport them over fiber, (and many PRIs are), it's a whole different setup from FiOS. And I'm really surprised that the audio industry would still prefer to use BRI...the whole 128kb of bandwidth isn't likely very "special" for bandwidth intensive flows, especially in this age of much higher throughput services being available. I do believe it's used more around live broadcast of voice, but other content would really suffer. We provide services to some radio stations that use APT WorldNet Oslo and Moseley Starlink codecs over TDM T1. These are to connect their transmitters to their studios, or interconnect studios. In one case, we deliver the T1 on fibre to their building, in another, we use an ILEC copper loop with PairGain HDSL2. They also still use ISDN BRI for backup. They use these circuits because they have no packet switching induced jitter, and extremely low latency. They came to us after being frustrated by the telco that kept wanting to sell them fibre ethernet, which wouldn't work with the equipment they needed to use. I pulled my last BRI line out of service 10 years ago. The older ATM based OLTs for GPON used to use a cell based protocol called BLES to provide a full POTS line emulation. The newer devices that lack an ATM bus, and perform the ATM to Ethernet conversion on each OLT card simply use IP. The ONT either uses SIP to go to a soft switch or SIP-PLAR to connect through GR-303 gateway to a legacy TDM switch. Our GPON deployment uses MGCP to our soft switch, as regular SIP doesn't provided the necessary protocol interaction to support all the functions needed for the 911 PSAP interconnection, whereas MGCP does. You could probably transport a T1 with PWE CES, then run it in to a channel bank to provide BRI service extension, but that's a whole lot of work to go through just for a 128k BRI. |
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rchandraStargate Universe fan Premium Member join:2000-11-09 14225-2105 ARRIS ONT1000GJ4 EnGenius EAP1250
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to HeadSpinning
Actually, it is theoretically possible to transport SIP over TCP/IP, but in my experience, far, far more often it is transported over UDP/IP. IIRC, SIPS is a different animal due to needing to exchange certificates and all that.
I also agree that despite the marketspeak and some of the tech people at Verizon calling it an analog service, I can't fathom it being analog except for the sole exception of the interface on the ONT. It would also be theoretically possible to vary the light beam intensity (AM) or color (FM) to actually transmit it over the fiber in analog fashion, but it's far more expedient to digitize the audio and use a digital transmission. Think of it if you will as simply taking an analog telephone adapter (ATA) and integrating it into the ONT (except the typical consumer ATA uses Ethernet). |
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rchandra |
to buckingham
That'd be "Session Initiation Protocol", BTW.
Live broadcast, yes, or any other application where constraining latency and jitter is an issue. An additional application would be live collaboration and recording, whether it be music or voiceovers or whatnot. Any transport which can offer those constraints would do, and the Internet unfortunately wouldn't qualify. Thus I was seeking a relatively low-cost way to get ISDN to me.
It looks more and more likely that without a SPECIFIC need (say, I already had a job arrangement), a home studio will just have to be put on hold. |
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buckinghamDoylstown Pa Premium Member join:2005-07-17 Buckingham, PA |
said by rchandra:That'd be "Session Initiation Protocol", BTW. Yup...fingers typed that without consulting brain... In the industry, we don't generally refer to IP phones as "digital phones". That term is normally used for TDM based digital sets (proprietary or ISDN), although it's certainly true that the transport used for IP sets (H.323 or SIP) is digital. |
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to rchandra
said by rchandra:Actually, it is theoretically possible to transport SIP over TCP/IP, but in my experience, far, far more often it is transported over UDP/IP. IIRC, SIPS is a different animal due to needing to exchange certificates and all that. SIP itself is transported of TCP/IP - the voice media is transported over RDP on UDP/IP. The SIP signaling goes between the SIP client and the call agent. The RTP traffic goes between the endpoints of the call, which may or may not be the same as the call agent. |
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rchandraStargate Universe fan Premium Member join:2000-11-09 14225-2105 |
rchandra
Premium Member
2013-Jul-22 8:29 pm
I guess we'll just have to disagree then. All the devices on my prem. are doing UDP when they do SIP.
emphasis in previous post on in my experience Apparently YMMV. |
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As per RFC 3261 section 18.1.1: If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. For most practical cases, your path MTU will almost certainly contain an Ethernet segment limited to 1500 bytes. With SIP requests getting more complicated, its pretty easy to hit that limit. I would suggest to you that in many deployments, TCP what's being used. The RTP streams are UDP. |
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rchandraStargate Universe fan Premium Member join:2000-11-09 14225-2105 ARRIS ONT1000GJ4 EnGenius EAP1250
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rchandra
Premium Member
2013-Jul-22 9:55 pm
said by HeadSpinning:As per RFC 3261 section 18.1.1: If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. For most practical cases, your path MTU will almost certainly contain an Ethernet segment limited to 1500 bytes. With SIP requests getting more complicated, its pretty easy to hit that limit. I would suggest to you that in many deployments, TCP what's being used. ...which still doesn't obviate the fact that while running Wireshark I've yet to see anything BUT UDP when I'm tracing what's going on. This is even with an instance of siproxd inserted, adding all sorts of things like Via: headers (thus making the packet(s) bigger). Somehow the 3 SIP clients here seem to keep all their traffic under that limit you mention. In fact just eyeballing a trace, looks like about 850 octets average. Again: TCP, possible. My experience, never used SIPS, and not seeing anything but UDP. ...just so we're clear. If by "many deployments" you mean SIPS, the exception of SIPS has been introduced from the start. Anything else, it just doesn't seem all that big. just sayin'...I've never questioned that, or even made a remark about RTP, so we have been in agreement on that one,. |
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While it is true that SIP signaling can run on both UDP and TCP (as per the RFC), it seems as if more complex applications are leaning towards TCP due to the size of the requests.
Regardless, I think we can all agree that both TCP/IP and UDP/IP are both IP technologies, and regardless of what NYCdave says, SIP, which is used by FDV IS VoIP, and not analogue. |
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buckinghamDoylstown Pa Premium Member join:2005-07-17 Buckingham, PA |
said by HeadSpinning:While it is true that SIP signaling can run on both UDP and TCP (as per the RFC), it seems as if more complex applications are leaning towards TCP due to the size of the requests. Agree...the large, enterprise SIP deployments I work with have always used TCP for signaling. |
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