I'm not sure whether "voice over LTE" (VoLTE) service, specifically, requires a data channel, but "voice over IP" (VoIP) does. If your voice calls, over a 4G/LTE network, sometimes (but not always) have an associated data channel, then my bet is, those are VoIP telephone calls, where the cell phone is using an IP network, to make the phone call. And note also, some of the non-technical literature you see, describing VoLTE, does not very clearly separate it, from VoIP. But they are very different services! Completely different mechanisms involved.

Parenthetically, my expectation would be that VoIP is the more normal way voice calls are made, over 4G/LTE cell networks. Because use of VoLTE requires that both sides of that voice connection are using 4G/LTE. So for example, even if both sides are using 4G/LTE smartphones, it's likely that at least one or the other is making that phone call over WiFi, as opposed to using their 4G channel. That too would bring in IP service.

When voice calls are made, from an IP network to the standard public switched telephone network (PSTN), or vice versa, there has to be a server involved, to map the PSTN telephone number with the IP address being used by the VoIP side of the connection. This is always needed, whether the call is going out on, or coming into, the IP network. In fact, it's also needed if both sides are in fact operating over IP, because both sides of that conversation will have used a standard telephone number to make the call, as opposed to having deliberately set up a TCP/IP session.

And it makes sense, right? The standard 10-digit telephone number, or other schemes used in other parts than North America, are not native to IP networks. Something has to do the translation, and something has to provide the steady stream of packets, required to keep that phone connection alive. Then you also have to set up the voice stream, between the IP-based end of the conversation, and the gateway to the PSTN circuit-switched side of the signal path. For example, you could be using Real Time Protocol (RTP), to carry the voice packets across. That too has associated data channel needs.

If you want to know the details, this RFC 3372 describes the Session Initiation Protocol (SIP) specifically as it is used for voice telephony, in the IP end of that phone connection.

https://www.rfc-editor.org/rfc/rfc3372.txt

SIP-T is not a new protocol - it is a set of mechanisms for interfacing traditional telephone signaling with SIP. The purpose of SIP-T is to provide protocol translation and feature transparency across points of PSTN-SIP interconnection. It intended for use where a VoIP network (a SIP network, for the purposes of this document) interfaces with the PSTN. Using SIP-T, there are three basic models for how calls interact with gateways. Calls that originate in the PSTN can traverse a gateway to terminate at a SIP endpoint, such as an IP phone. Conversely, an IP phone can make a call that traverses a gateway to terminate in the PSTN. Finally, an IP network using SIP may serve as a transit network between gateways - a call may originate and terminate in the PSTN, but cross a SIP-based network somewhere in the middle.

There is a more basic RFC to cover SIP, in general, not necessarily related to telephone calls made over IP networks. RFC 3261:

https://www.rfc-editor.org/rfc/rfc3261.txt

There are many applications of the Internet that require the creation and management of a session, where a session is considered an exchange of data between an association of participants. The implementation of these applications is complicated by the practices of participants: users may move between endpoints, they may be addressable by multiple names, and they may communicate in several different media - sometimes simultaneously. Numerous protocols have been authored that carry various forms of real-time multimedia session data such as voice, video, or text messages. The Session Initiation Protocol (SIP) works in concert with these protocols by enabling Internet endpoints (called user agents) to discover one another and to agree on a characterization of a session they would like to share. For locating prospective session participants, and for other functions, SIP enables the creation of an infrastructure of network hosts (called proxy servers) to which user agents can send registrations, invitations to sessions, and other requests. SIP is an agile, general-purpose tool for creating, modifying, and terminating sessions that works independently of underlying transport protocols and without dependency on the type of session that is being established.

This is excellent Albert, because LTE is based on the IETF standards and when calls are routed to the Packet Data Gateway to be connected they must follow that standard. Let me research and re-read your post and the associated RFC. I didn't follow IETF too much, but it's incorporated in 4G and the 3GPP standards.

(We always thought those IEFT guys were "cowboys".

VoLTE does operate over a data channel, but this is a separate EPS Bearer than those used for regular data channels. The VoLTE bearer is 'owned' and managed by the by the network operator rather and typically will have a higher quality of service to try to ensure that the voice experience more closely matches that we expect from circuit switched voice used by 2G, 3G and the 'plain old telephone service'.

Thank you, Tim. I assume you mean that the packet network is of higher quality than the general "Public Internet" so the operator can more carefully control the traffic on that network and allow higher QoS?

Albert, In reading your referenced it says that SIP is an application layer protocol and essentially is the signaling protocol for call set up and tear down. This makes sense that this would be the data channel. (Yes, I know, all of LTE is data.)

Tim, it seems that you are saying that VoIP and VoLTE are not the same. VoLTE uses a higher QoS bearer, but the bear is none the less IP. So I am wondering how VoLTE and VOIP are different?

Then back to my original question. Is the carriers using one IP bearer for signaling, (likely SIP) and a second one for the voice bearer? Again, there needs to be transmit and receive bearer so 1 IP address or 3?

Albert, thanks for the nice explanation. Both 'regular' VOIP and VoLTE use SIP and are in many ways similar. VoLTE allows control of access channel feature not available to the ordinary VOIP - examples are 'header compression' and TTI bundling - both of which make more efficient use of the radio channel bandwidth.

If your LTE Network Operator offers VoLTE then you will be able to call anyone-anywhere by dialling their phone number from your cellphone exactly in the same way that you would using Circuit-switched voice on GSM or WCDMA. In LTE if you place a call and you don't have VoLTE then the network will place the call using Circuit-Switch Fallback (CSFB) providing that there is a GSM or 3G network available to fall back to. If you don't have VoLTE and there is no 2G/3G coverage then you won't be able to make a call in that way. Typically you will have to place the call using a separate application, and depending on the application you may or may not be able to call subscribers on other networks. An example is Skype, where you can call any Skype subscriber, but you need credit to pay for a local telephone call to deliver your call to a POTS telephone.

Ben, the Data Bearer and the VoLTE bearer will be associated with different IP Addresses. VoLTE only uses one IP Address. Signalling is not carried on a separate IP Address - I think you can consider this to be in-band signalling.

Excellent, I see from reading some of your references that if you are LTE the circuit switched fall back is often IMS and there is an interworking required between the core (EPC) and those networks.

But what is interesting from your post, Tim, is that the Data Bearer and the VoLTE bearer have different IP addresses. But VoLTE signaling (SIP?) is in-band essentially. So what's the Bearer left to do? Is that simply the default PDN connection?

Hi Ben, CSFB will not be IMS, it will be classical circuit switch vocoded speech - and will be re-routed to wherever it needs to go back at the switch. So any IMS connection will be torn down - it may need to be 're-built' at the switch if the connected party was connected by IMS.

You can do VOIP over a 3G bearer, but it won't be VoLTE (as it won't be over LTE).

You can do VOIP over a GPRS/EGPRS bearer but in my experience call quality would be (politely put) 'variable'.

So this data connection is for what? When the phone leaves IMS and needs to connect to 3G?

Hi Ben, This all goes back to GSM/GPRS and a service call Dual Transfer Mode (DTM) which enabled simultaneous Voice and user Data Transmission which was a significant enhancement to GPRS - so much so that initial implementations suspended the data while a voice channel was connected - before that you could not start voice while a data connection was made - or vice versa. Anything like this was almost unthinkable in the 1980s & 90's!) - we were connecting to internet using 14.4 kbps dial-in modems!. Early 3G phones (& networks) had similar limitations. However, when HS-Data came along then integration of voice and data was 'de rigeur'.

In VoLTE voice and data are completely simultaneous, but are on different bearers, which gives the network operator better control of QoS for the voice channel. If you leave an area where VoLTE is unsupported - or indeed go into a region where there is no LTE coverage you will be transferred to another service - typically to WCDMA, and the VoLTE channel will be moved on to a Circuit Switched Voice Channel. If your voice channel is being carried by a non-VoLTE OTT VOIP service and you are moved to a 3G service this will just be moved to a 3G data service carrying your VOIP speech.

Excellent. I've got to go through the standards on this.

I remember putting "voice priority" in 3GPP2 standards. DTM must have been thought of during ISDN which had 2 voice bearers and a data bearer simultaneously (2B + D).

Hi Ben, Yes, so I recall, but here X21/X25 phones never got much traction, even in the business world, DTM came quite a time after ISDN, essentially in the second or third wave of GSM handsets, late 1990s and early 2000's, not long before 3GPP/3GPP2 started to emerge from the development labs..

On standards ... me too! The problem is that the 3GPP standards now take up many yards (or meters) of shelf space! and several years of life ploughing your way through them.

Hi Tim,

I was reading this thread

you mentioned "the Data Bearer and the VoLTE bearer will be associated with different IP Addresses." But i couldn't find two ip address in my LTE phone settings that has both VolTE network and data browsing capability. Any idea?

I'm looking at the charging data records (CDRs). With each voice call, there is a matching data sessions that starts exactly 1 second prior to the call and ends approximately at the end of the call. It appears to be the call-associated signaling channel.