oracle 8 linux. rtpengine 11.2 with uek kernel
There are some specific TIPS for ORACLE linux 8 and UEK core for compiling\installing rtpengine.
for test vanilla server i will use oracle 8 linux:
take ISO from: https://yum.oracle.com/oracle-linux-isos.html
set http repository: http://yum.oracle.com/repo/OracleLinux/OL8/baseos/latest/x86_64
INSTALL RTPENGINE 11.2.2.0
git clone https://bitbucket.org/yooxy/centos8-rtpengine10-all-codecs.git
cd centos8-rtpengine10-all-codecs
#use precompiled pkgs from RPMS/el8/ and RPMS/el8/11 dirs
dnf -y install epel-release
dnf config-manager --set-enabled ol8_codeready_builder
dnf -y install --nogpgcheck https://download1.rpmfusion.org/free/el/rpmfusion-free-release-8.noarch.rpm
dnf -y install SDL2
dnf install kernel-uek-devel
cd RPMS/el8
dnf localinstall ffmpeg-libs-4.2.7-1.el8.x86_64.rpm libavdevice-4.2.7-1.el8.x86_64.rpm
cd 11
dnf localinstall ngcp-rtpengine-11.2.2.0+0~mr11.2.2.0-1.el8.x86_64.rpm ngcp-rtpengine-kernel-11.2.2.0+0~mr11.2.2.0-1.el8.x86_64.rpm ngcp-rtpengine-dkms-11.2.2.0+0~mr11.2.2.0-1.el8.noarch.rpm
INSTALL UEK KERNEL
We will using UEK 5.4.17. check “uname -a” maybe you already have 5.4.17 kernel, if not:
dnf config-manager --disable ol8_UEKR7 dnf config-manager --enable ol8_UEKR6 dnf install kernel-uek kernel-uek-devel reboot
SIGN MODULE ORACLE 8
Issue “Key was rejected by service” may happens if you have enabled “Secure boot”.
To check it: run “mokutils –sb-state”. In enabled state you have to sign your rtpengine module for kernel. To add your key do this:
mokutil –import /var/lib/dkms/mok.pub
enter password you wish, and after reboot you see in BOIS invitation to enroll your keys. do it with password you have entered before and then xt_rtpengine module will be loaded correctly.
ISSUES:
If you have installed UEK 5.15+ then you have to update you GCC compiler to 11+ or you will have error when compiling kernel module for rtpengine: you are using 11.5 GCC for compiling kernel and 8.5 for compiling xt_rtpengine.
modprobe: FATAL: Module xt_RTPENGINE not found in directory
Extension RTPENGINE revision 0 not supported, missing kernel module?
ERR: [core] FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED
dkms (kernel) module xt_RTPENGINE not compiled at all.
in common cases you have to install kernel-uek-devel-<version>, where version is “uname -r”
../lib/codeclib.h:53:10: fatal error: libswresample/swresample.h: No such file or directory
dnf install libswresample-devel -y
/lib/codeclib.h:54:10: fatal error: libavcodec/avcodec.h: No such file or directory
dnf install libavcodec-devel -y
./include/media_player.h:29:10: fatal error: libavformat/avformat.h: No such file or directory
dnf install libavformat-devel -y
FAILED TO OPEN/DELETE KERNEL TABLE 0 (Permission denied), KERNEL FORWARDING DISABLED
rtpengine kernel module named xt_RTPENGINE creates 2 file in /proc/rtpengine, control and list. If you see permission of this files it will be only for root. That is why rtpengine can not use as ngcp-rtpengine.
Ofcourse somewhere good solution is present, but you can just change ngcp-rtpengine to root in rtpengine.services file and everything will start correctly.
RTPENGINE cluster TIPS
There main concept here: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications
but few important thing to check:
1. Redis have disabled keyspace notification to enable change to notify-keyspace-events “AKE” in redis.conf
2. interface names should start exactly from “pub”
3. If you have not active interface IP on passive (pub2) rtpengine you have to set sysctl net.ipv4.ip_nonlocal_bind=1
Tags: cluster, foreign sessions, not working, rtpengine
| Posted in opensips, rtpengine | No Comments »
7.10.2022 | Posted in opensips, rtpengine | No Comments »
Rtpengine. Opus. Ilbc.
I am using ilbc to make calls with mobile applications. As we know ilbc is old codec, all tests,table and pictures all over the net make us feel as ilbc most worse then opus. because opus is faster, more quality e.t.c.
Seems… my opinion is different, if you want good quality and minimum bandwidth out of box then use ilbc – it’s not problem, it will have 23kb bandwidth for one side. Audio bandwidth up to 4kHz so voice will be good enough for conversation. But there is no way to reduce bitrate and bandwidth with ilbc, so let’s try to implement opus.
also, converting 48kHz(sample rate) files with libopus is slowest then with ilbc in most of cases. Be noted that you can not use opus with 8kHz(sample rate) in default configuration by rtpengine only 48kHz is supported.
Components used for testing:
- centos 7: iftop, tcpdump
- custom ffmpeg 4.2.7,
- opus 1.3.2,
- opus-tools: opusenc, opusrtp
- rtpengine 11.0.1.5,
- microsip 3,
- sipp 3.6.
- clumsy ( simulate bad network on windows)
I will start from end, maybe it will be helpful for someone. What was my aim, i wanted to use packet loss, fec, speech mode, low bandwidth from opus codec.
Success:
* low bandwidth with 8kb\s bitrate (13kb\s actual) and 40ms frame duration.
Failed:
* packet_loss (no way to understand if it really works, real test does not show that it helps)
* fec, ( same as packet loss)
* speech mode, (take more cpu resources when encoding without real result)
How to add opus support into rtpengine.
For encoding\decoding opus rtpeginge using ffmpeg library. so you have to be sure that libopus is present with ffmpeg. you can do that with: “ffmpeg -h encoder=libopus” if you don’t see: “Codec ‘libopus’ is not recognized by FFmpeg.” then seems ffmpeg have opus with libopus encoder\decoder.
How to set parameters for opus codec:
when you do rtpengine_offer use this: as one of params:
codec-transcode-opus/48000/2/8000/40/maxaveragebitrate–8000;maxplaybackrate–12000;useinbandfec–1;ptime–40;maxptime–40/ar-48000,b–8000
where is :
48000 – sample rate in SDP
2 – channels (default for opus)
8000 (b\s) – bitrate for codec implement on encoder side
40 (ms) – frame duration (should affect on encoder side, but you have a=ptime 20 in SDP, codec will work on 20 ms)
“maxaveragebitrate–8000;maxplaybackrate–12000;useinbandfec–1;ptime–40;maxptime–40” there are a=fmtp parameters into SDP, you can check what it means in RFC.
“ar–48000,b–8000” – codec individual options you can take a look ffmpeg docs to check what you can use. For some reason individual options for opus codec like packet_loss can not be set by this logic, you have to set it inside codeclib.c in rtpengine source code . for example “
if (enc->ptime > 0 ) {
codeclib_set_av_opt_int(enc, "frame_duration", enc->ptime);
codeclib_set_av_opt_int(enc, "packet_loss", 5);
codeclib_set_av_opt_int(enc, "fec", 1);
codeclib_set_av_opt_int(enc, "application", 2048);
}
issues: when you set 40 ms frame_duration for opus and you have not any a=ptime 40 in SDP towards to destination peer, peer will not send stream with 40 ms frame_duration, maybe there is a bug into Microsip. Using ptime and maxptime into codec options – not helps.
so, to avoid this i did add ptime=40 as rtpengine_offer parameter and add little fix to codeclib.c to make 40ms default ptime for opus codec.
How to check speed of converting with libopus
you need any music input file, for example any.wav. then you may try to use
“ffmpeg -i madonna-48k.wav -c libopus -ab 18000 madonna.opus“
it will convert wav file to opus with bitrate 18k\s
as result you will see some data ended with
“size= 82kB time=00:00:39.83 bitrate= 16.8kbits/s speed= 136x”
also you can convert it with libilbc encoder:
“ffmpeg -i madonna-48k.wav -c libilbc -ar 8000 -ab 18000 madonna.lbc“
you will see:
“size= 74kB time=00:00:39.84 bitrate= 15.2kbits/s speed= 170x“
to be continued….
30.10.2021Talant Blogs about VOIP
Tags: asterisk, expert, Kamailio, rtpengine, voip
| Posted in Asterisk, kamailio, rtpengine, Эксперт | No Comments »
1.10.2021 | Posted in Asterisk, kamailio, rtpengine, Эксперт | No Comments »
RTPENGINE DTMF transcoding
Kamailio:
route[rtpengine_invite] {
if (has_body("application/sdp"))
rtpengine_manage("codec-mask=telephone-event transcode=PCMA always-transcode");
}
route[rtpengine_answer] {
if (has_body("application/sdp"))
rtpengine_manage("always-transcode");
}
Demo for transcoding from telephone-event – to in-band and vice versa
Full kamailio.cfg
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v5.2 - default configuration script
# - web: https://www.kamailio.org
# - git: https://github.com/kamailio/kamailio
#
# Direct your questions about this file to: <sr-users@lists.kamailio.org>
#
# Refer to the Core CookBook at https://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Note: the comments can be:
# - lines starting with #, but not the pre-processor directives,
# which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef,
# #!subst, #!substdef, ...
# - lines starting with //
# - blocks enclosed in between /* */
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
# - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
# block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To block 401 and 407 authentication replies execute:
# - define WITH_BLOCK401407
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
####### Include Local Config If Exists #########
import_file "kamailio-local.cfg"
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
/* number of SIP routing processes */
children=8
/* uncomment the next line to disable TCP (default on) */
# disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
* based on reverse DNS on IPs (default on) */
auto_aliases=yes
/* add local domain aliases */
# alias="sip.mydomain.com"
/* uncomment and configure the following line if you want Kamailio to
* bind on a specific interface/port/proto (default bind on all available) */
listen=udp:192.168.1.199:5090
#!ifdef WITH_TLS
enable_tls=yes
#!endif
/* life time of TCP connection when there is no traffic
* - a bit higher than registration expires to cope with UA behind NAT */
tcp_connection_lifetime=3605
####### Custom Parameters #########
/* These parameters can be modified runtime via RPC interface
* - see the documentation of 'cfg_rpc' module.
*
* Format: group.id = value 'desc' description
* Access: $sel(cfg_get.group.id) or @cfg_get.group.id */
#!ifdef WITH_PSTN
/* PSTN GW Routing
*
* - pstn.gw_ip: valid IP or hostname as string value, example:
* pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
*
* - by default is empty to avoid misrouting */
pstn.gw_ip = "" desc "PSTN GW Address"
pstn.gw_port = "" desc "PSTN GW Port"
#!endif
#!ifdef WITH_VOICEMAIL
/* VoiceMail Routing on offline, busy or no answer
*
* - by default Voicemail server IP is empty to avoid misrouting */
voicemail.srv_ip = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
####### Modules Section ########
/* set paths to location of modules */
# mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223")
loadmodule "dialog.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")
# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# set next param to 1 to add value to ;lr param (helps with some UAs)
modparam("rr", "enable_full_lr", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
# modparam("registrar", "max_contacts", 10)
/* max value for expires of registrations */
modparam("registrar", "max_expires", 3600)
/* set it to 1 to enable GRUU */
modparam("registrar", "gruu_enabled", 0)
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
* if you enable this parameter, be sure the enable "append_fromtag"
* in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
/* register callback to match myself condition with domains list */
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
/* ip ban htable with autoexpire after 5 minutes */
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
modparam("debugger", "log_level_name", "exec")
#!endif
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
dlg_manage();
route(rtpengine_invite);
setflag(FLT_ACC); # do accounting
}
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
}
# Wrapper for relaying requests
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood detection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself) {
if($sht(ipban=>$si)!=$null) {
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
exit;
}
if (!pike_check_req()) {
xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif
if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(is_method("OPTIONS") && uri==myself && $rU==$null) {
sl_send_reply("200","Keepalive");
exit;
}
if(!sanity_check("17895", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;
if (has_body("application/sdp"))
route(rtpengine_invite);
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
} else if ( is_method("ACK") ) {
# ACK is forwarded statelessly
route(NATMANAGE);
} else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
}
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
exit;
}
# Handle SIP registrations
route[REGISTRAR] {
if (!is_method("REGISTER")) return;
if(isflagset(FLT_NATS)) {
setbflag(FLB_NATB);
#!ifdef WITH_NATSIPPING
# do SIP NAT pinging
setbflag(FLB_NATSIPPING);
#!endif
}
if (!save("location")) {
sl_reply_error();
}
exit;
}
# User location service
route[LOCATION] {
#!ifdef WITH_SPEEDDIAL
# search for short dialing - 2-digit extension
if($rU=~"^[0-9][0-9]$") {
if(sd_lookup("speed_dial")) {
route(SIPOUT);
}
}
#!endif
#!ifdef WITH_ALIASDB
# search in DB-based aliases
if(alias_db_lookup("dbaliases")) {
route(SIPOUT);
}
#!endif
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
if (is_method("INVITE")) {
setflag(FLT_ACCMISSED);
}
route(RELAY);
exit;
}
# Presence server processing
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE")) return;
if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {
route(TOVOICEMAIL);
# returns here if no voicemail server is configured
sl_send_reply("404", "No voicemail service");
exit;
}
#!ifdef WITH_PRESENCE
if (!t_newtran()) {
sl_reply_error();
exit;
}
if(is_method("PUBLISH")) {
handle_publish();
t_release();
} else if(is_method("SUBSCRIBE")) {
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null) {
sl_send_reply("404", "Not here");
exit;
}
return;
}
# IP authorization and user authentication
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address()) {
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself) {
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself) {
sl_send_reply("403","Not relaying");
exit;
}
#!else
# authentication not enabled - do not relay at all to foreign networks
# if(uri!=myself) {
# sl_send_reply("403","Not relaying");
# exit;
# }
#!endif
return;
}
# Caller NAT detection
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop()) {
set_contact_alias();
}
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control and signaling updates for NAT traversal
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;
if(nat_uac_test("8")) {
rtpproxy_manage("co");
} else {
rtpproxy_manage("cor");
}
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}
# URI update for dialog requests
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (uri==myself) return;
append_hf("P-hint: outbound\r\n");
route(RELAY);
exit;
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n");
return;
}
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return;
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
# normalize target number for pstn gateway
# - convert leading 00 to +
if (starts_with("$rU", "00")) {
strip(2);
prefix("+");
}
if (strempty($sel(cfg_get.pstn.gw_port))) {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
} else {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
+ $sel(cfg_get.pstn.gw_port);
}
route(RELAY);
exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
# Routing to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE")) return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");
return;
}
if(is_method("INVITE")) {
if($avp(oexten)==$null) return;
$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
} else {
if($rU==$null) return;
$ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
}
route(RELAY);
exit;
#!endif
return;
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]") {
route(rtpengine_answer);
route(NATMANAGE);
}
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) exit;
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_BLOCK401407
# block call redirect based on 401, 407 replies.
if (t_check_status("401|407")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
$du = $null;
route(TOVOICEMAIL);
exit;
}
#!endif
}
route[rtpengine_invite] {
if (has_body("application/sdp"))
rtpengine_manage("codec-mask=telephone-event transcode=PCMA always-transcode");
}
route[rtpengine_answer] {
if (has_body("application/sdp"))
rtpengine_manage("always-transcode");
}
28.09.2021 kamailio rtpengine media timeout TIPs
- timeout will be raised only if both sides of rtp is silent
- you have to enable tcp at kamailio config (disable_tcp=no)
- you must have in kamailio cfg: listen=tcp:127.0.0.1:8090, loadmodule “xmlrpc.so”, and additional params:
loadmodule "xmlrpc.so"
modparam("xmlrpc", "route", "XMLRPCS")
modparam("xmlrpc", "url_skip", "^/sip")
modparam("xmlrpc", "url_match", "^/RPC2")
4. you have to add route XMLRPC:
route[XMLRPCS] {
xlog("L_ALERT","RPC recieved");
dispatch_rpc();
}
5. config rtpengine should have:
b2b-url = http://127.0.0.1:8090/RPC2 xmlrpc-format = 2
6. after restart kamailio and rtpengine you may to check you may do command from command line:
curl http://127.0.0.1:8090/RPC2
output should be like that:
<?xml version="1.0"?> <methodResponse> <fault> <value> <struct> <member> <name>faultCode</name> <value><int>400</int></value> </member> <member> <name>faultString</name> <value><string>Invalid XML-RPC Document</string></value> </member> </struct> </value> </fault> </methodResponse>29.09.2020
rtpengine-installation-configuration
установка rtpengine На debian 10. проходит на ура, вместе с g729 кодеком, т.е. можно использовать transonding.
исходная статья.
репозиторий со скриптами: https://bitbucket.org/yooxy/rtpengine-debian-10-install/src/master/
скрипт для debian 10 buster:
#!/usr/bin/sh apt update apt install devscripts python3-debian equivs git -y #for some reason my debian take old version libsystemd-dev by default apt install libsystemd-dev=247.3-6~bpo10+1 -y git clone https://github.com/sipwise/rtpengine.git #there are some steps to make fix for buster OS cd rtpengine/pkg/deb bash ./generator.sh bash ./backports/buster cp -r buster/* ../../debian cd ../../ #install dependencies in automatically way mk-build-deps --install #compile rtpengine dpkg-buildpackage -b -us -uc cd ../ dpkg -i rtpengine-daemon_11.4.0.0+0~mr11.4.0.0_amd64.deb rtpengine-iptables_11.4.0.0+0~mr11.4.0.0_amd64.deb rtpengine-kernel-dkms_11.4.0.0+0~mr11.4.0.0_all.deb rtpengine-utils_11.4.0.0+0~mr11.4.0.0_all.deb rtpengine_11.4.0.0+0~mr11.4.0.0_all.deb apt --fix-broken install
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