[L] Change in libosmo-netif[master]: beginning of twrtp documentation

falconia has uploaded this change for review. ( https://gerrit.osmocom.org/c/libosmo-netif/+/39291?usp=email ) Change subject: beginning of twrtp documentation ...................................................................... beginning of twrtp documentation Change-Id: I90976edfd8c66c2d1c5bc7939e0a49f725a02f3f --- M Makefile.am M configure.ac A doc/Makefile.am A doc/manuals/Makefile.am A doc/manuals/twrtp-guide-docinfo.xml A doc/manuals/twrtp-guide.adoc 6 files changed, 267 insertions(+), 1 deletion(-) git pull ssh://gerrit.osmocom.org:29418/libosmo-netif refs/changes/91/39291/1 diff --git a/Makefile.am b/Makefile.am index fd03b74..defd38e 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,7 +1,7 @@ AUTOMAKE_OPTIONS = foreign dist-bzip2 1.6 ACLOCAL_AMFLAGS = -I m4 -SUBDIRS = include src examples utils tests +SUBDIRS = include src doc examples utils tests pkgconfigdir = $(libdir)/pkgconfig pkgconfig_DATA = libosmo-netif.pc diff --git a/configure.ac b/configure.ac index 9b52901..8ee5a4d 100644 --- a/configure.ac +++ b/configure.ac @@ -126,12 +126,56 @@ AC_PATH_PROG(DOXYGEN,doxygen,false) AM_CONDITIONAL(HAVE_DOXYGEN, test $DOXYGEN != false && test "x$doxygen" = "xyes") +# Generate manuals +AC_ARG_ENABLE(manuals, + [AS_HELP_STRING( + [--enable-manuals], + [Generate manual PDFs [default=no]], + )], + [osmo_ac_build_manuals=$enableval], [osmo_ac_build_manuals="no"]) +AM_CONDITIONAL([BUILD_MANUALS], [test x"$osmo_ac_build_manuals" = x"yes"]) +AC_ARG_VAR(OSMO_GSM_MANUALS_DIR, [path to common osmo-gsm-manuals files, overriding pkg-config and "../osmo-gsm-manuals" + fallback]) +if test x"$osmo_ac_build_manuals" = x"yes" +then + # Find OSMO_GSM_MANUALS_DIR (env, pkg-conf, fallback) + if test -n "$OSMO_GSM_MANUALS_DIR"; then + echo "checking for OSMO_GSM_MANUALS_DIR... $OSMO_GSM_MANUALS_DIR (from env)" + else + OSMO_GSM_MANUALS_DIR="$($PKG_CONFIG osmo-gsm-manuals --variable=osmogsmmanualsdir 2>/dev/null)" + if test -n "$OSMO_GSM_MANUALS_DIR"; then + echo "checking for OSMO_GSM_MANUALS_DIR... $OSMO_GSM_MANUALS_DIR (from pkg-conf)" + else + OSMO_GSM_MANUALS_DIR="../osmo-gsm-manuals" + echo "checking for OSMO_GSM_MANUALS_DIR... $OSMO_GSM_MANUALS_DIR (fallback)" + fi + fi + if ! test -d "$OSMO_GSM_MANUALS_DIR"; then + AC_MSG_ERROR("OSMO_GSM_MANUALS_DIR does not exist! Install osmo-gsm-manuals or set OSMO_GSM_MANUALS_DIR.") + fi + + # Find and run check-depends + CHECK_DEPENDS="$OSMO_GSM_MANUALS_DIR/check-depends.sh" + if ! test -x "$CHECK_DEPENDS"; then + CHECK_DEPENDS="osmo-gsm-manuals-check-depends" + fi + if ! $CHECK_DEPENDS; then + AC_MSG_ERROR("missing dependencies for --enable-manuals") + fi + + # Put in Makefile with absolute path + OSMO_GSM_MANUALS_DIR="$(realpath "$OSMO_GSM_MANUALS_DIR")" + AC_SUBST([OSMO_GSM_MANUALS_DIR]) +fi + AC_OUTPUT( libosmo-netif.pc include/Makefile include/osmocom/Makefile include/osmocom/netif/Makefile src/Makefile + doc/Makefile + doc/manuals/Makefile examples/Makefile utils/Makefile tests/Makefile diff --git a/doc/Makefile.am b/doc/Makefile.am new file mode 100644 index 0000000..f2c4de1 --- /dev/null +++ b/doc/Makefile.am @@ -0,0 +1 @@ +SUBDIRS = manuals diff --git a/doc/manuals/Makefile.am b/doc/manuals/Makefile.am new file mode 100644 index 0000000..800d0e1 --- /dev/null +++ b/doc/manuals/Makefile.am @@ -0,0 +1,9 @@ +EXTRA_DIST = twrtp-guide-docinfo.xml twrtp-guide.adoc + +if BUILD_MANUALS + ASCIIDOC = twrtp-guide.adoc + include $(OSMO_GSM_MANUALS_DIR)/build/Makefile.asciidoc.inc + + OSMO_REPOSITORY = libosmo-netif + include $(OSMO_GSM_MANUALS_DIR)/build/Makefile.common.inc +endif diff --git a/doc/manuals/twrtp-guide-docinfo.xml b/doc/manuals/twrtp-guide-docinfo.xml new file mode 100644 index 0000000..3537c23 --- /dev/null +++ b/doc/manuals/twrtp-guide-docinfo.xml @@ -0,0 +1,22 @@ + +<authorgroup> + <author> + <firstname>Mychaela</firstname> + <surname>Falconia</surname> + &lt;email&gt;falcon(a)freecalypso.org&lt;/email&gt; + <authorinitials>MNF</authorinitials> + <affiliation> + <shortaffil>SD2G</shortaffil> + <orgname>San Diego 2G Association</orgname> + <jobtitle>Founder and CTO</jobtitle> + </affiliation> + </author> +</authorgroup> + +<legalnotice> + <para> + The author of this document is a stateless person who does not + participate in the institution of copyright. This document + is in the public domain. + </para> +</legalnotice> diff --git a/doc/manuals/twrtp-guide.adoc b/doc/manuals/twrtp-guide.adoc new file mode 100644 index 0000000..2d1dce4 --- /dev/null +++ b/doc/manuals/twrtp-guide.adoc @@ -0,0 +1,190 @@ += Guide to ThemWi RTP endpoint library + +The RTP implementation library described in this document was developed by +Mother Mychaela N. Falconia under Themyscira Wireless branding and then +contributed to Osmocom in 2025. It is currently under consideration for +merging into `libosmo-netif`; once merged, it will be automatically available +to any application that is built with the standard suite of Osmocom libraries. + +This document is written to describe the library as it will appear and +function if the proposal to merge it into `libosmo-netif` is accepted. + +== Introduction + +ThemWi RTP endpoint library, consisting of `<osmocom/netif/twrtp.h>` and +`<osmocom/netif/twjit.h>` blocks provided by `libosmo-netif`, is intended +for interworking between an RTP stream and a fixed timing system such as +GSM Um interface TCH or T1/E1 TDM. Such interworking consists of two +fundamental elements: + +* In every fixed time quantum, the interworking element receives a unit of + speech or CSData media from the fixed source and emits an RTP packet + carrying that quantum of speech or CSData. + +* In the opposite direction, the fixed timing system requires a quantum of + speech or CSData to be fed to it on every tick without fail, yet the + interworking element has no control over when RTP packets may arrive + from the IP network. This direction of interworking requires a rather + complex element called a jitter buffer, an element whose design and + configuration always involves some trade-offs and compromises. + +=== Domain of application + +The present library is *not* intended to be an all-purpose implementation +of IETF RFCs 3550 and 3551, supporting all possible RTP use cases as +envisioned by IETF. Instead it is intended to support RTP _as it is used_ +in these two specific telecom network environments: + +* 3GPP networks that use RTP according to TS 26.102 and TS 48.103; + +* The way RTP is used to transport G.711 PSTN traffic across the public + Internet in what may be colloquially referred to as IP-PSTN. + +The two 3GPP specs referenced above prescribe a fixed packetization time +of 20 ms for all codecs on AoIP interface. Furthermore, they stipulate +that: + +* In the case of compressed speech transport, each RTP packet carries + exactly one frame of the speech codec in use; + +* In the case of uncompressed G.711 speech or CSData transport, each RTP + packet carries exactly 160 payload octets (20 ms worth) of what would + have been a 64 kbit/s timeslot in T1/E1 transport. + +This fixed-quantum property, namely the property that every RTP packet +carries exactly one fixed quantum of speech or CSData, where the duration +of this quantum is known at connection setup time and cannot suddenly +change from one packet to the next, is required by the present ThemWi +RTP endpoint library - this requirement constitutes a fundamental aspect +of its architectural design. + +An RTP endpoint implementation library that imposes the just-described +requirement is sufficient for the purpose of building IP-based GSM networks +that follow 3GPP TS 48.103 (the requirements of that spec are in agreement +with the library constraint), and it is also sufficient for interfacing +to IP-PSTN by way of common commercial PSTN-via-SIP connectivity providers. + +In the case of IP-PSTN, the author of the present library has experience +only with North American PSTN-via-SIP connectivity providers. In all +of our operational experience so far, these IP-PSTN connectivity providers +behave in ways that are fully compatible with the expectations of the +present RTP library, as long as the following conditions are met: + +* No attempt is made to use any codecs other than PCMU or PCMA: + don't include any other codecs in the SDP offer, and send only SDP answers + that select either PCMU or PCMA out of the received multi-codec offer. + +* No attempt is made to use any other `ptime` besides the most common + industry standard of 20 ms. + +In all operational experience so far, incoming INVITE SDPs indicate either +`a=ptime:20` or `a=maxptime:20`, and when we indicate `a=ptime:20` in all +SDPs we send out, the IP-PSTN peer always sends us 20 ms RTP packets, +as opposed to some other packetization interval which would break +the fixed-quantum model assumed by the present RTP library. + +However, it needs to be acknowledged that the present library is *not* +suitable for general-purpose, IETF-style applications outside of +``walled garden'' 3GPP networks or the semi-walled environment of +IP-PSTN with ``well-behaved'' entities: there are many behaviors that +are perfectly legal per the RFCs, but are not supported by the present +library. Having a peer send RTP with a packetization interval that is +different from what we asked for via `ptime` attribute is one of those +behaviors that is allowed by IETF, but not supported by this library. + +=== Configurable quantum duration and time scale + +For every RTP stream it handles, the library needs to know two key +parameters: + +* The scale or ``clock rate'' used for RTP timestamps, i.e., how many + timestamp units equal one millisecond of physical time; + +* The ``quantum'' duration in milliseconds. + +*Quantum* is the term used in this RTP endpoint library for the unit +of speech or CSData carried in one RTP packet. In Kantian philosophy terms, +a quantum of speech or CSData is the thing-in-itself (a single codec frame, +or a contiguous chunk of 160 PCM samples grabbed from an ISDN B channel), +whereas the RTP packet that carries said quantum is one particular transport +representation of that thing-in-itself. + +In most applications of this library (all 3GPP codecs other than AMR-WB, +and all IP-PSTN applications in our experience so far), the time scale +is 8000 timestamp units per second (or 8 per millisecond, as it appears +in the actual APIs) and the time duration of a single quantum is 20 ms, +hence one quantum equals 160 timestamp units. +Both parameters (RTP timestamp clock rate in kHz and the number of ms +per quantum) are configurable at the time of endpoint creation, allowing +RTP endpoints for AMR-WB, or perhaps G.711 or CSData applications with +different packetization times - but they cannot be changed later in +the lifetime of an allocated endpoint. + +For ease of exposition, the rest of this document will assume that +one quantum equals 20 ms in time or 160 RTP timestamp units. If these +numbers are different in your application, substitute accordingly. + +== Jitter buffer model + +In the interworking direction from incoming RTP to the fixed timing system, +the latter will poll the RTP endpoint (or more precisely, the jitter buffer +portion thereof) for a quantum of media every 20 ms, whenever that quantum +is required for transmission on GSM Um TCH or for TDM output etc. +The job of the jitter buffer is to match previously received RTP packets +to these fixed-timing output polls, while striving to meet these two +conflicting goals: + +* Any time that elapses between an RTP packet being received and its + payload being passed as a quantum to the fixed timing system constitutes + added latency - which needs to be minimized. + +* IP-based transport always involves some jitter: the time delta between + the receipt of one RTP packet and the arrival of its successor is very + unlikely to be exactly equal to 20 ms every time. This jitter may be + already present in the RTP stream from its source if that source is + an IP-native BTS that does not pass through E1 Abis and thus exposes + the inherent jitter of GSM TDMA multiframe structure, but even if + the source is perfectly timed, some jitter will still be seen on the + receiving end. Depending on the actual amount of jitter seen in a + given deployment, it may be necessary to introduce some latency-adding + buffering in the receiving RTP endpoint - otherwise the function of + interworking to the fixed timing system at the destination will perform + poorly, as will be seen in the ensuing sections. + +This chapter covers the design and operation of `twjit`, the jitter buffer +component of ThemWi RTP endpoint library. + +=== Flows and handovers + +In `twjit` terminology, a single RTP flow is a portion (or the whole) +of an RTP stream that exhibits the following two key properties: + +* All packets have the same SSRC; + +* The RTP timestamp increment from each packet to the next always equals + the fixed quantum duration expressed in timestamp units, i.e., 160 + in most practical applications. + +A handover in `twjit` terminology is a point in the incoming RTP stream +at which either of the following events occurs: + +* An SSRC change is seen; + +* The RTP timestamp advances by an increment that is not an integral multiple + of the expected fixed quantum duration. In contrast, if an RTP timestamp + increment is seen that _is_ an integral multiple of the quantum, but that + integral multiple is more than one, and there is enough buffering in the + system such that this event is seen before the jitter buffer underruns, + such events are *not* treated as handovers: instead it is assumed to be + an occurrence of either packet loss or reordering. + +A handover in `twjit` is thus a transition from one flow to the next. +This term was adopted because such transitions are expected to occur +when an RTP stream belonging to a single call switches from one BSS endpoint +to another (in the same BSS or in a different one) upon radio handover events +in GSM and other cellular networks, but handovers in `twjit` sense can also +occur in other applications that aren't GSM. For example, if an IP-PSTN peer +we are conversing with suddenly decides, for its own reasons known only to +itself, to change its SSRC or jump its RTP output timescale, our `twjit` +instance will treat that event as a handover. + -- To view, visit https://gerrit.osmocom.org/c/libosmo-netif/+/39291?usp=email To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings?usp=email Gerrit-MessageType: newchange Gerrit-Project: libosmo-netif Gerrit-Branch: master Gerrit-Change-Id: I90976edfd8c66c2d1c5bc7939e0a49f725a02f3f Gerrit-Change-Number: 39291 Gerrit-PatchSet: 1 Gerrit-Owner: falconia &lt;falcon(a)freecalypso.org&gt;