laforge has uploaded this change for review. ( https://gerrit.osmocom.org/c/pysim/+/39292?usp=email )
Change subject: cards: Avoid exception seen with (some) GSM-R SIM cards
......................................................................
cards: Avoid exception seen with (some) GSM-R SIM cards
Some old cards are classic SIM and not based on UICCs. Such cards
do not offer the capability of selecting applications. Let's avoid
running into an exception by providing dummy methods that simply fail
for each AID selection.
Change-Id: Ib3457496380c0c5096052ad7799970ee620dee33
Closes: OS#6691
---
M pySim/cards.py
1 file changed, 10 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/pysim refs/changes/92/39292/1
diff --git a/pySim/cards.py b/pySim/cards.py
index b7958f4..4fde211 100644
--- a/pySim/cards.py
+++ b/pySim/cards.py
@@ -73,6 +73,16 @@
# callers having to do hasattr('read_aids') ahead of every call.
return []
+ def adf_present(self, adf: str = "usim") -> bool:
+ # a non-UICC doesn't have any applications. Convenience helper to avoid
+ # callers having to do hasattr('read_aids') ahead of every call.
+ return False
+
+ def select_adf_by_aid(self, adf: str = "usim", scc: Optional[SimCardCommands] = None) -> Tuple[Optional[Hexstr], Optional[SwHexstr]]:
+ # a non-UICC doesn't have any applications. Convenience helper to avoid
+ # callers having to do hasattr('read_aids') ahead of every call.
+ return (None, None)
+
class SimCardBase(CardBase):
"""Here we only add methods for commands specified in TS 51.011, without
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Hello Jenkins Builder,
I'd like you to reexamine a change. Please visit
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Change subject: nokia_site: introduce hopping control for Nokia *Site
......................................................................
nokia_site: introduce hopping control for Nokia *Site
nokia_site hopping-type 0 sets baseband hopping
nokia_site hopping-type 1 sets sythesizer (RF) hopping
Signed-off-by: csaba.sipos <metro4(a)freemail.hu>
Change-Id: Idb1d269d2ee80d72ede2394125e8acedf0ee8b06
---
A doc/examples/osmo-bsc/nokia/osmo-bsc.metrosite.RF-hopping.2TRX.cfg
M include/osmocom/bsc/bts.h
M src/osmo-bsc/bts_nokia_site.c
M src/osmo-bsc/bts_vty.c
4 files changed, 209 insertions(+), 6 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-bsc refs/changes/90/39290/4
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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>
+ <email>falcon(a)freecalypso.org</email>
+ <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.
+
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Change subject: nokia_site: introduce hopping control for Nokia *Site
......................................................................
Patch Set 3:
(1 comment)
Patchset:
PS3:
@laforge@osmocom.org
Jenkins gave this error:
Documentation error (missing docs):
<command id='nokia_site nokia_hopping (0|1)'>
<param name='1' doc='(null)' />
Not sure how and where to fix this. Do I need to create a new example config file in examples/nokia, or I can add it to the existing one, maybe I need to add this somewhere else? Looked at a few bts_vty.c additions lately, but none of them edited anything else.
The patch itself is runtime tested against InSite (no hopping), and MetroSite (RF hopping), both works as expected.
Thanks!
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Hello Jenkins Builder,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/c/osmo-bsc/+/39290?usp=email
to look at the new patch set (#3).
The following approvals got outdated and were removed:
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Change subject: nokia_site: introduce hopping control for Nokia *Site
......................................................................
nokia_site: introduce hopping control for Nokia *Site
nokia_site hopping-type 0 sets baseband hopping
nokia_site hopping-type 1 sets sythesizer (RF) hopping
Signed-off-by: csaba.sipos <metro4(a)freemail.hu>
Change-Id: Idb1d269d2ee80d72ede2394125e8acedf0ee8b06
---
M include/osmocom/bsc/bts.h
M src/osmo-bsc/bts_nokia_site.c
M src/osmo-bsc/bts_vty.c
3 files changed, 49 insertions(+), 6 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-bsc refs/changes/90/39290/3
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Change subject: nokia_site: introduce hopping control for Nokia *Site
......................................................................
Patch Set 2:
(1 comment)
File src/osmo-bsc/bts_vty.c:
Robot Comment from checkpatch (run ID jenkins-gerrit-lint-20869):
https://gerrit.osmocom.org/c/osmo-bsc/+/39290/comment/7715478d_8db87d3d?usp… :
PS2, Line 478: if (!is_nokia_bts(bts)) {
suspect code indent for conditional statements (16, 32)
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Hello Jenkins Builder,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/c/osmo-bsc/+/39290?usp=email
to look at the new patch set (#2).
The following approvals got outdated and were removed:
Verified-1 by Jenkins Builder
Change subject: nokia_site: introduce hopping control for Nokia *Site
......................................................................
nokia_site: introduce hopping control for Nokia *Site
nokia_site hopping-type 0 sets baseband hopping
nokia_site hopping-type 1 sets sythesizer (RF) hopping
Signed-off-by: csaba.sipos <metro4(a)freemail.hu>
Change-Id: Idb1d269d2ee80d72ede2394125e8acedf0ee8b06
---
M include/osmocom/bsc/bts.h
M src/osmo-bsc/bts_nokia_site.c
M src/osmo-bsc/bts_vty.c
3 files changed, 49 insertions(+), 6 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-bsc refs/changes/90/39290/2
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Gerrit-Change-Number: 39290
Gerrit-PatchSet: 2
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laforge has posted comments on this change by pespin. ( https://gerrit.osmocom.org/c/osmo-pcap/+/39288?usp=email )
Change subject: client: Introduce osmo_pcap_file to prepare support for pcapng
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Patch Set 1:
(1 comment)
File src/osmo_pcap_file.c:
https://gerrit.osmocom.org/c/osmo-pcap/+/39288/comment/350e4c6d_f6f85c9c?us… :
PS1, Line 38: osmo_pcap_file_global_header_size
do we reeally need a function for this? Do we ever expect that there will be a future file format which will return anything but a constant here? Especially since we are not passing any data/context into the function, I have a hard time imagining any future implementation that would not return a constant?
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Gerrit-Change-Id: Ifd389a2a32897e101c62fd280ddca984d485d373
Gerrit-Change-Number: 39288
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Gerrit-Comment-Date: Sat, 11 Jan 2025 14:56:19 +0000
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