Opensips Notes

16.08.2022

Oracle Centos 8. Rtpengine with all codecs supported.

As result of this instruction you will have all this codecs supported in your centos 8 installations.

                PCMA: fully supported
                PCMU: fully supported
                G723: fully supported
                G722: fully supported
                G729: fully supported
               G729a: fully supported
               speex: fully supported
                 GSM: fully supported
                iLBC: fully supported
                opus: fully supported
                 AMR: fully supported
              AMR-WB: fully supported
     telephone-event: fully supported
                  CN: fully supported

Synopsis:

RPMS, build and install scripts: git clone https://bitbucket.org/yooxy/centos8-rtpengine10-all-codecs.git

This instruction will give you RTPENGINE for Centos 7 and Centos 8 withh all codecs. RPM packages in RPMS dir are ready for install. But also you have rpmbuild-rtpengine.el7 and rpmbuild-rtpengine.el8 to compile it on your system in automatically way.

if you start to compiling on new system, then everything should go fine after you type sh rpmbuild-rtpengine.el7.

IF you work on production system , then check files you are running before start due to you may to install unnecessary packets or kernels. 

To build rtpengine with all codecs (g729,AMR,opus,iLBC, GSM) on Centos 8:

cd ~
git clone https://bitbucket.org/yooxy/centos8-rtpengine10-all-codecs.git
sh rpmbuild-rtpengine.el8
cd ~/rpmbuild/RPMS/
dnf install noarch/ngcp-rtpengine-dkms-10.5.1.3+0~mr10.5.1.3-1.el8.noarch.rpm x86_64/ngcp-rtpengine-kernel-10.5.1.3+0~mr10.5.1.3-1.el8.x86_64.rpm x86_64/ngcp-rtpengine-10.5.1.3+0~mr10.5.1.3-1.el8.x86_64.rpm

Your RPMs ready for install in ~/root/rpmbuild/RPMS

---

To install rtpengine without build 10.5 run “sh install-rtpengine.el7”

10.12.2021

Ansible Part II. Install opensips,opensips-cli,opensips control panel.

You can use this Ansible roles to install full Opensips + Control Panel with one command.

For some reason roles to install Opensips from ansible galaxy not working as expected.

I have modified some roles to make it works.

This will good only for Debian 10 and Centos 7. Most popular systems. 

Roles will install Mysql server with defaults, opensips-cli, opensips 3.2 and opensips control panel with opensips DB. Access to panel is login “admin” and password “opensips”.

Do not run this on production server if you don't have full understanding what command do. It may cause your system loose some important things like ssh keys.

1. Make usr/local/bin inpath to run command from it.
   export PATH=$PATH:/usr/local/sbin
   echo “export PATH=$PATH:/usr/local/sbin” > /root/.bashrc
2. Install ansible on debian 10
   apt install git python-pip
   pip install ansible
3. Generate SSH key for control node host (it should be present in authorized_hosts file on every managed nodes)
   ssh-keygen -t rsa -b 4096
4. Get repository with modified roles
   git clone https://bitbucket.org/yooxy/ansible-opensips.git
5. Put roles into /root/.ansible repository.
   mkdir /root/.ansible
   mkdir /root/.ansible/roles
   cp -r ansible-opensips/roles /root/.ansible
6. Modify hosts file in ansible-opensips repo and copy it to /etc/ansible (default directory for hosts file)
   Run ansible-playbook where you have “ansible-playbook inst_opensips.yml”

Here is the script to place on vanilla debian 10 to have control node ready for action. Just do step 6 after this script done.

export PATH=$PATH:/usr/local/sbin
echo "export PATH=$PATH:/usr/local/sbin" > /root/.bashrc
apt install git python-pip -y
pip install ansible 
ssh-keygen -t rsa -b 4096
git clone https://bitbucket.org/yooxy/ansible-opensips.git
mkdir /etc/ansible
mkdir /root/.ansible
mkdir /root/.ansible/roles
cp -r ansible-opensips/roles /root/.ansible

20.09.2021

RTPENGINE + G729 DEBIAN 10.10 and Debian 11

apt update
apt upgrade -y
apt install -y linux-headers-$(uname -r) linux-image-$(uname -r)

##reboot

mkdir /opt/rtpengine

cd /opt/rtpengine

apt install -qqy git curl mariadb-server libmosquitto-dev libwebsockets-dev python3-websockets apt-utils devscripts dpkg-dev debhelper iptables iptables-dev libcurl4-openssl-dev libglib2.0-dev libavcodec-extra libhiredis-dev libpcre3-dev libssl-dev libxmlrpc-core-c3-dev markdown zlib1g-dev module-assistant dkms gettext default-libmysqlclient-dev gperf libavcodec-dev libavfilter-dev libavformat-dev libavutil-dev libbencode-perl libcrypt-openssl-rsa-perl libcrypt-rijndael-perl libdigest-crc-perl libdigest-hmac-perl libevent-dev libio-multiplex-perl libio-socket-inet6-perl libjson-glib-dev libnet-interface-perl libpcap0.8-dev libsocket6-perl libswresample-dev libsystemd-dev nfs-common netcat-openbsd netcat unzip libconfig-tiny-perl libspandsp-dev**

git clone https://github.com/sipwise/rtpengine.git .
cp etc/rtpengine.sample.conf /etc/rtpengine/rtpengine.conf

VER=1.0.4

curl https://codeload.github.com/BelledonneCommunications/bcg729/tar.gz/$VER >bcg729_$VER.orig.tar.gz

tar zxf bcg729_$VER.orig.tar.gz

cd bcg729-$VER

git clone https://github.com/ossobv/bcg729-deb.git debian

dpkg-buildpackage -us -uc -sa

cd ../

dpkg -i libbcg729-*.deb


export DEBIAN_FRONTEND=noninteractive

apt-get update -qqy

mkdir -p ./debian/flavors

touch ./debian/flavors/no_ngcp


dpkg-checkbuilddeps

dpkg-buildpackage -b -us -uc

dpkg -i ../*.deb

**manual edit: /etc/rtpengine/rtpengine.conf: set your IPs.

If you want to add webrtc2sip feature use next lines

certbot certonly -d webrtc.domain.com

**manual edit: kamailiorc: uncomment db engine string

kamdbctl create kamailio 


download template kamailio.cfg, tls.cfg and kamailio_local.cfg 

(git clone https://bitbucket.org/erewin/webrtc2sip-template.git)

systemctl start rtpengine
systemctl start kamailio

To make the same on Debian 11

#!/usr/bin/sh
apt update
apt upgrade

#here you have reboot

#!/usr/bin/sh
mkdir /opt/rtpengine

cd /opt/rtpengine

apt-get install git curl -y

apt-get install libmosquitto-dev libwebsockets-dev python3-websockets apt-utils dpkg-dev debhelper iptables libxtables-dev libip6tc-dev libip4tc-dev libcurl4-openssl-dev libglib2.0-dev libavcodec-extra libhiredis-dev libpcre3-dev libssl-dev libxmlrpc-core-c3-dev markdown zlib1g-dev module-assistant dkms gettext default-libmysqlclient-dev gperf libavcodec-dev libavfilter-dev libavformat-dev libavutil-dev libbencode-perl libcrypt-openssl-rsa-perl libcrypt-rijndael-perl libdigest-crc-perl libdigest-hmac-perl libevent-dev libio-multiplex-perl libio-socket-inet6-perl libjson-glib-dev libnet-interface-perl libpcap0.8-dev libsocket6-perl libswresample-dev libsystemd-dev nfs-common netcat-openbsd netcat unzip libconfig-tiny-perl libspandsp-dev
#debian 11
apt install libxtables-dev libip6tc-dev libip4tc-dev libiptc-dev

apt install -y linux-headers-$(uname -r) linux-image-$(uname -r)

git clone https://github.com/sipwise/rtpengine.git .


VER=1.0.4

curl https://codeload.github.com/BelledonneCommunications/bcg729/tar.gz/$VER >bcg729_$VER.orig.tar.gz

tar zxf bcg729_$VER.orig.tar.gz

cd bcg729-$VER

git clone https://github.com/ossobv/bcg729-deb.git debian

dpkg-buildpackage -us -uc -sa

cd ../

dpkg -i libbcg729-*.deb


export DEBIAN_FRONTEND=noninteractive

apt-get update -qqy

mkdir -p ./debian/flavors

touch ./debian/flavors/no_ngcp


dpkg-checkbuilddeps

dpkg-buildpackage -b -us -uc

dpkg -i ../*.deb

8.08.2021

Kamailio TLS-UDP with dispatcher and used old openssl 1.0.2 (sslv2\3 support)

This project is how to convert TLS-UDP with kamailio. Problem is that modern Unix (Ubuntu and debian have only modern openssl library so it’s not support ssv2\3 protocol). To make it works you have to

• do this steps at vanilla system.
• get source of kamailio
• for compiling using /make include_modules=”tls”/
• rtpengine should be installed as usual
• then everything should go as usual

Benefits from this configs is that working for inbound\outbound calls and use Dispatcher.
this is example of dispatcher file:

#
# dispatcher destination sets (groups)
#

# line format
# setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt)


1 sip:PEER1:5070;transport=udp 0 0 socket=udp:KAM_SOCKET_UDP:5070
1 sip:PERR2;transport=udp 0 0 socket=udp:KAM_SOcKET_UDP:5070

2 sip:TLS_CARRIER:5061;transport=tls 0 0 socket=tls:TLS_SOCKET_KAM:5061

Example of kamailio config you may get here.

#!KAMAILIO
#
############################################################
# *** Value defines - IDs used later in config
#!ifdef WITH_DEBUG
#!define DBGLEVEL 3
#!else
#!define DBGLEVEL 2
#!endif
#!define DS_LIST "/usr/local/etc/kamailio/dispatcher.list"
#!define LISTEN_UDP_PRIVATE udp:LOCAL_INTERFACE_IP:5070
#!define FLAG_FROM_ASTERISK 10
#!define FLAG_FROM_PEER 11


# - flags
#   FLT_ - per transaction (message) flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5

#	FLB_ - per branch flags
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

####### Global Parameters #########

/* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */
debug=DBGLEVEL

/* set to 'yes' to print log messages to terminal or use '-E' cli option */
log_stderror=no

memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "

children=8

auto_aliases=no

server_signature=no

alias="DOMAIN_NAME"

listen=udp:LOCAL_INTERFACE_IP:5070
listen=tls:LOCAL_INTERFACE_IP:5061 advertise EXTERNAL_IP:5061

tcp_connection_lifetime=3605

tcp_max_connections=20000

tcp_accept_no_cl=yes

enable_tls=yes
tls_max_connections=20000

enable_sctp=no

####### Modules Section ########
mpath="/usr/local/lib64/kamailio/modules/"


#loadmodule "db_mysql.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "pike.so"
loadmodule "htable.so"
loadmodule "nathelper.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
#loadmodule "registrar"
#loadmodule "usrloc.so"
loadmodule "textops.so"
loadmodule "textopsx.so"
loadmodule "dialog.so"
loadmodule "tls.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
loadmodule "dispatcher.so"
loadmodule "outbound.so"
loadmodule "rtpengine.so"
loadmodule "path.so"


####Module Specific Parameters####
modparam("rr", "enable_double_rr", 1)

modparam("tls", "config", "/etc/kamailio/tls.cfg")
modparam("path", "use_received", 1)


modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")

modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223")
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")


modparam("dispatcher", "list_file", DS_LIST)
#modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "ds_ping_interval", 20)
modparam("dispatcher", "ds_ping_from", "sip:keepalive@smsglobal.com")
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("nathelper", "received_avp", "$avp(s:rcv)")


###Routing Logic

request_route {

	# per request initial checks
	route(REQINIT);
	
	# 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();
	}

        route(CHECK_SOURCE_IP);

	# handle requests within SIP dialogs
	route(WITHINDLG);

	### only initial requests (no To tag)

	# 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")) {
		setflag(FLT_ACC); # do accounting
	}

	# 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
	route(DISPATCH);
}


route[RELAY] {
    if (is_method("INVITE")) {
        if(!t_is_set("failure_route")) {
            t_on_failure("MANAGE_FAILURE");
        }
    }

    if (isflagset(FLAG_FROM_PEER)) {
	    xlog("L_INFO","seems call from $si goig from PEER");
	    
    } else {
            
            xlog("L_INFO","Relaying TO TLS\n ");
    }


    if (!t_relay()) {
            sl_reply_error();
    }
    #exit;

}

# Per SIP request initial checks
route[REQINIT] {
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}

	if(!sanity_check("1511", "7")) {
		xlog("Malformed SIP message from $si:$sp\n");
		exit;
	}
	force_rport();
	
	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;
		}
	}

	if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") {
		# silent drop for scanners - uncomment next line if want to reply
		# sl_send_reply("200", "OK");
		exit;
	}
	if(is_method("OPTIONS") && uri==myself && $rU==$null) {
		sl_send_reply("200","Keepalive");
		exit;
	}



}

route[CHECK_SOURCE_IP] {
    if(ds_is_from_list("1")) {
            setflag(FLAG_FROM_ASTERISK);
    } else {
            setflag(FLAG_FROM_PEER);
    }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
	if (has_totag()) {
		# sequential request withing a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("BYE")) {
				setflag(FLT_ACC); # do accounting ...
				setflag(FLT_ACCFAILED); # ... even if the transaction fails
			}
			route(RELAY);
		} else {
			if (is_method("SUBSCRIBE") && uri == myself) {
				# in-dialog subscribe requests
				exit;
			}
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# non loose-route, but stateful ACK;
					# must be ACK after a 487 or e.g. 404 from upstream server
					t_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;

	sl_send_reply("404", "Not Acceptable");
	exit;
}

# Presence server route
route[PRESENCE] {
	if(!is_method("PUBLISH|SUBSCRIBE"))
		return;

	sl_send_reply("404", "Not Acceptable");
	exit;
}

# Dispatch requests
route[DISPATCH] {
        # round robin dispatching on gateways group '1'
        # record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|REFER")) {
                record_route();
                if (has_body("application/sdp")) {
                    if (rtpengine_offer()) {
                            t_on_reply("1");
                    }
                } else {
                    t_on_reply("2");
                }

                if (isflagset(FLAG_FROM_PEER)) {
		    xlog("L_INFO","Call from $si seems from PEER");
                    if(!ds_select_domain("1", "4")) {
                            send_reply("404", "No destination");
                            exit;
                    }

                }

                if (isflagset(FLAG_FROM_ASTERISK)) {
		    xlog("L_INFO","Call from $si seems from ASTERISK");
                    if(!ds_select_domain("2", "4")) {

                            send_reply("404", "No destination");
                            exit;
                    }
		    xlog("L_INFO","Call from $si seems from ASTERISK [$du] [$ru]");
		    
                }
		    

		    xlog("L_INFO","DESTINATION is $du");
        }

        if (is_method("ACK") && has_body("application/sdp")) {
                rtpengine_answer("force");
        }

        route(RELAY);
}


failure_route[RTF_DISPATCH] {
	if (t_is_canceled()) {
		exit;
	}
	# next DST - only for 500 or local timeout
	if (t_check_status("500")
			or (t_branch_timeout() and !t_branch_replied())) {
		if(ds_next_dst()) {
			xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n");
			t_on_failure("RTF_DISPATCH");
			route(RELAY);
			exit;
		}
	}
}

onreply_route[1] {
        if (has_body("application/sdp")) {
                rtpengine_answer("force");
        }
}

onreply_route[2] {
        if (has_body("application/sdp")) {
                rtpengine_offer("force");
        }
}

31.07.2021

install opensips 3.1 Debian 10 + RTPPROXY 2.2

apt update
apt upgrade -y
apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B
echo "deb https://apt.opensips.org buster 3.1-releases" >/etc/apt/sources.list.d/opensips.list
echo "deb https://apt.opensips.org buster cli-nightly" >/etc/apt/sources.list.d/opensips-cli.list
apt update
apt install opensips*
apt install mariadb-server
opensips-cli -> database create opensips
apt install build-essential letsencrypt -y
(rtpproxy manual: https://www.rtpproxy.org/doc/master/user_manual.html#idm650)
useradd rtpproxy
cd /usr/src
git clone -b master https://github.com/sippy/rtpproxy.git
git -C rtpproxy submodule update --init --recursive
cd rtpproxy
./configure
make clean all
make install
put this content to /lib/systemd/system/rtpproxy.service 
-----
[Unit]
Description=RTPProxy media server
After=network.target
Requires=network.target

[Service]
Type=simple
PIDFile=/var/run/rtpproxy/rtpproxy.pid
Environment='OPTIONS= -f -L 4096 -l 0.0.0.0 -m 10000 -M 20000 -d INFO:LOG_LOCAL5'

Restart=always
RestartSec=5

ExecStartPre=-/bin/mkdir /var/run/rtpproxy
ExecStartPre=-/bin/chown rtpproxy:rtpproxy /var/run/rtpproxy

ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp:127.0.0.1:22222 \
 -u rtpproxy:rtpproxy -n udp:127.0.0.1:22223 $OPTIONS

ExecStop=/usr/bin/pkill -F /var/run/rtpproxy/rtpproxy.pid

ExecStopPost=-/bin/rm -R /var/run/rtpproxy

StandardOutput=syslog
StandardError=syslog
SyslogIdentifier=rtpproxy
SyslogFacility=local5

TimeoutStartSec=10
TimeoutStopSec=10

[Install]
WantedBy=multi-user.target
---------------------------

30.07.2021

Front End.

Для создания телефонных проектов частно нужен веб интерфейс, скажем сейчас я даже не знаю ни одного проекта без предоставления веб-интерфейса. Все знают что есть фреймворки, и простые вещи такие как показать таблички из одной базы, что-то загрузить что-то отфильтровать – просто. Сложности начинаются когда надо действительно внедрить что-то что вы не знаете как делать силами фреймворка, а знать фреймворк хорошо это быть полноценным веб разработчиком. На этом этапе нам не нужно быть веб-разработчиками, но мы должны давать клиента хорошего качества вебстраницу на которой он сможет управлять своими телефонными сервисами.

Итак, я остановился, на том, что в проектах буду использовать:

• PHP 7.3+
• SMARTY 3+
• BOOTSTRAP 4 + готовый шаблон (для админки например)
• JS
• JQUERY + МОДУЛИ
• CHARTS.JS

Это простые системы, хорошо задокументированные. При помощи них легко создавать страницы для управления таблицами, загрузками или выгрузкой файлов и даже рисовать графики.

А шаблонизатор smarty позволяет разделить код и дизайн, чтобы фрон-энд разработчик в дальнейшем смог, делать любой дизайн.

Примеры:

26.07.2021

To deploy some Voip service

• What server you want to use:
  • Standalone (physically server)
  • Dedicated Server
  • Virtual Machine (AWS,GCP,DO and so on)
• Do you have requirements for OS and related software:
  • Debian\Centos\Ubuntu
  • Mysql/PostgreSQL
• What Voip trunk you have or how you have connected to voice services:
  • I have two trunks with a-z routes and cheap prices.
• What amount of calls you expect
  • normal: less 5cps, less: 100 concurrent channels,
  • high load: less 20 cps, less 1000 cc,
  • highest unlimited cps, unlimited cc,

22.04.2021

Аренда серверов со скидками

Мы предоставляем в аренду полноценные виртуальные АТС (freepbx\Issabel\FusionPBX) по запросу на электронный адрес.

• для тестов 500 рублей\мес. (2 cpu\2 гб\20Гб HDD\external ip\domain 3 level),
• для работы 1500 рублей\мес. (2 cpu\4 гб\60Гб HDD\external ip\domain 3 level),
• для работы 3500 рублей\мес. (4 cpu\4 гб\120Гб HDD\external ip\domain 2 level),

Скидки у партнеров: для краткосрочной аренды и проведение тестов самое оно!

12.03.2021

183 ringback. OPensips + Rtpproxy.

Задача подменить КПВ идущее от провайдера на 10 секунд, далее прокидывать КПВ от провайдера. Задача решается при помощи opensips 3.1 + rtpproxy 2.1 и минимальной конфигурацией.

rtpproxy прекрасно компилируется на базу Ubuntu 20
opensips 3.1 устанавливается из репозиториев с конфигурацией по умолчанию
для проигрывания своего приветсвия необходимо сделать 2 вещи:

а) подключить rtpproxy и использовать rtpproxy_stream2uac функцию. Пример находится в репозитории.

б) для rtpproxy приветствия необходимо подготовить в нужном формате при помощи makeann (создается после компиляции).

репозиторий с примером конфига.

Схема решения:

client -> opensips -> carrier.

22.12.2020

opensips as SBC

Example of opensips SBC with 2 interfaces with full RTP proxy and g729 transcoding.

Software: opensips 3.1, rtpengine, bcg729.

example settings for rtpengine:

OPTIONS="-i external/172.18.254.50!EXTERNAL_IP -i internal/172.25.150.242 -n 127.0.0.1:2223 -m 35000 -M 65000 -L 4 --log-facility=local1 --table=0 --delete-delay=0 --timeout=60 --silent-timeout=600 --final-timeout=7200 –offer-timeout=60 --num-threads=4 --tos=184 –no-fallback"

example opensips config: git clone https://bitbucket.org/yooxy/opensips-sbc-local-external-transcode.git

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