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b2b: add refer notify cancelled scenarios
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@razvancrainea
razvancrainea committed Sep 20, 2023
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20 changes: 20 additions & 0 deletions README.md
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Expand Up @@ -361,6 +361,26 @@ Run the same as
scenario, but the referee no longer sends the re-INVITE, thus we retry
the bridging after 2.1-4s.

#### 31.refer-unattended-uac-notify-cancel
Run a transfer similar to
[09.refer-unattended-uac-notify-fail](#09refer-unattended-uac-notify-fail), but
the call times out and the referee's call gets canceled.

#### 32.refer-unattended-uas-notify-cancel
Run a transfer similar to
[10.refer-unattended-uas-notify-fail](#10refer-unattended-uas-notify-fail), but
the call times out and the referee's call gets canceled.

#### 33.refer-unattended-uac-notify-terminate
Run a transfer similar to
[09.refer-unattended-uac-notify-fail](#09refer-unattended-uac-notify-fail), but
the remaining participant terminates the call during transfer.

#### 34.refer-unattended-uas-notify-terminate
Run a transfer similar to
[10.refer-unattended-uas-notify-fail](#10refer-unattended-uas-notify-fail), but
the remaining participant terminates the call during transfer.

### UAC Auth

Verifies the behavior of uac_auth module.
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3 changes: 3 additions & 0 deletions b2b/31.refer-unattended-uac-notify-cancel/defines.yml
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---
transfer_ip: 192.168.52.5
transfer_port: 5065
113 changes: 113 additions & 0 deletions b2b/31.refer-unattended-uac-notify-cancel/opensips.cfg
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#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions <[email protected]>
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# https://opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########
######################################################################
/* uncomment the following lines to enable debugging */
#debug_mode=yes

log_level=4
xlog_level=4
log_stderror=yes

udp_workers=4

####### Modules Section ########

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 3)
modparam("tm", "fr_inv_timeout", 5)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### B2B modules
loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_req_route", "b2b_logic_request")

loadmodule "proto_udp.so"

####### Routing Logic ########

# main request routing logic

route {

if (!mf_process_maxfwd_header(10)) {
send_reply(483,"Too Many Hops");
exit;
}

if (has_totag()) {
send_reply(481, "Call/Transaction Does Not Exist");
exit;
}

# accept just INVITE requests
if (!is_method("INVITE")) {
send_reply(503, "Service Unavailable");
exit;
}

# create the server entity
b2b_server_new("caller");
# create the initial client entity, to connect the caller with the callee
b2b_client_new("callee", $ru);

# initialize B2B session for the "refer" scenario
b2b_init_request("refer");

exit;
}

route[b2b_logic_request] {
if ($rm != "REFER") {
# for requests other than REFER, no special actions needs to be done,
# just pass the request to the peer
b2b_pass_request();
exit;
}

# end dialog with the referrer
b2b_send_reply(202, "Accepted");

# create the client entity corresponding to
# the user specified in the 'Refer-To' header
b2b_client_new("referee", $hdr(Refer-To));

# bridge the referrer's peer with the referee
b2b_bridge("peer", "referee", ,"notify, no-late-sdp, rollback-failed");
}
37 changes: 37 additions & 0 deletions b2b/31.refer-unattended-uac-notify-cancel/scenario.yml
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---
timeout: 20

tasks:
- name: OpenSIPS
type: opensips

- name: SIPP UAS
type: uas-sipp
config_file: scripts/uas.xml
require: OpenSIPS

- name: SIPP UAC
type: uac-sipp
config_file: scripts/uac.xml
duration: 2000 # before and after transfer
username: {{ username }}
remote: {{ uas_ip }}:{{ uas_port }}
require:
started:
task: SIPP UAS
wait: 0.5
keys:
transfer: sip:transfer@{{ transfer_ip }}:{{ transfer_port }}

- name: SIPP Transfer
type: uas-sipp
config_file: scripts/transfer.xml
ip: {{ transfer_ip }}
port: {{ transfer_port }}
require:
- started:
task: SIPP UAC
wait: 0.5
- after:
task: OpenSIPS
wait: 0.5
157 changes: 157 additions & 0 deletions b2b/31.refer-unattended-uac-notify-cancel/scripts/transfer.xml
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->

<scenario name="Basic UAS responder">
<nop>
<action>
<assignstr assign_to="domain" value="[domain]"/>
</action>
</nop>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv request="INVITE" crlf="true" rrs="true">
<action>
<!-- there should only be one Via -->
<ereg regexp=".+" search_in="hdr" header="Via:" check_it="true" assign_to="dummy"/>
<ereg regexp="," search_in="hdr" header="Via:" check_it_inverse="true" assign_to="dummy"/>
<ereg regexp=".+" search_in="hdr" header="Via:" occurence="2" check_it_inverse="true" assign_to="dummy"/>
<!-- there should only be no Route or Record-Route headers -->
<ereg regexp=".+" search_in="hdr" header="Route:" check_it_inverse="true" assign_to="dummy"/>
<!-- check the contact is the proxy -->
<ereg regexp="<sip:([^;@]*)>" search_in="hdr" check_it="true" header="Contact:" assign_to="dummy,contact" />
<strcmp assign_to="result" variable="contact" variable2="domain"/>
<ereg regexp="." search_in="var" check_it_inverse="true" variable="result" assign_to="dummy"/>
<ereg regexp="[:space:]*B2B." search_in="hdr" check_it="true" header="Call-ID:" assign_to="dummy" />
<ereg regexp=".*" search_in="hdr" header="Via" assign_to="remote_via"/>
<ereg regexp=".*" search_in="hdr" header="CSeq" assign_to="remote_cseq"/>
</action>
</recv>

<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[service].[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>

<pause milliseconds="100"/>

<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[service].[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>

<recv request="CANCEL"
rtd="true"
crlf="true">
<action>
<!-- there should only be one Via -->
<ereg regexp=".+" search_in="hdr" header="Via:" check_it="true" assign_to="dummy"/>
<ereg regexp="," search_in="hdr" header="Via:" check_it_inverse="true" assign_to="dummy"/>
<ereg regexp=".+" search_in="hdr" header="Via:" occurence="2" check_it_inverse="true" assign_to="dummy"/>
<!-- there should only be no Route or Record-Route headers -->
<ereg regexp=".+" search_in="hdr" header="Route:" check_it_inverse="true" assign_to="dummy"/>
<ereg regexp="[:space:]*B2B." search_in="hdr" check_it="true" header="Call-ID:" assign_to="dummy" />
</action>
</recv>


<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[service].[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>

<send>
<![CDATA[
SIP/2.0 487 Request Terimnated
Via[$remote_via]
[last_From:]
[last_To:];tag=[service].[call_number]
[last_Call-ID:]
CSeq[$remote_cseq]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>


<recv request="ACK"
rtd="true"
crlf="true">
<action>
<!-- there should only be one Via -->
<ereg regexp=".+" search_in="hdr" header="Via:" check_it="true" assign_to="dummy"/>
<ereg regexp="," search_in="hdr" header="Via:" check_it_inverse="true" assign_to="dummy"/>
<ereg regexp=".+" search_in="hdr" header="Via:" occurence="2" check_it_inverse="true" assign_to="dummy"/>
<!-- there should only be no Route or Record-Route headers -->
<ereg regexp=".+" search_in="hdr" header="Route:" check_it_inverse="true" assign_to="dummy"/>
<ereg regexp="[:space:]*B2B." search_in="hdr" check_it="true" header="Call-ID:" assign_to="dummy" />
</action>
</recv>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->

<!-- <pause milliseconds="10"/> -->

<!-- Pause 8 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<!-- <pause milliseconds="8000"/> -->

<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="200, 500"/>

<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500, 1000, 2000, 3000"/>

<Reference variables="dummy,domain,contact,result"/>

</scenario>
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