%block text-align=left% '''Author: Liviu Chircu <liviu@opensips.org>'''

%block text-align=left% '''written by Liviu Chircu <liviu@opensips.org>'''

!Voice Transcoding in OpenSIPS using Sangoma D-series Cards
%block text-align=right% '''written by Liviu Chircu'''

%red%As mentioned in the {+[[http://www.opensips.org/html/docs/modules/2.2.x/sngtc.html|module documentation]]+}, transcoding may only be done if the UAC performs early SDP negotiation and the UAS supports late SDP negotiation.%%

%red%As mentioned in the {+[[http://www.opensips.org/html/docs/modules/2.2.x/sngtc.html|module documentation]]+}, transcoding may only be done if the UAC performs early SDP negotiation.%%

%red%As mentioned in the {+[[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]]+}, transcoding may only be done if the UAC performs early SDP negotiation.%%

This tutorial illustrates the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client. The latest version of the transcoding library, as of 1 August 2013 is: '''sng-tc-linux-1.3.4.1'''

http://www.opensips.org/images/tc-setup.png

(:allVersions Tutorials-SangomaVoiceTranscoding 1.11 :)

(:versionLine Tutorials-SangomaVoiceTranscoding 1.11 :)

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(:includeHTMLxc http://www.yahoo.com border=3 scroll=yes:)

(:includeHTML http://www.yahoo.com border=3 scroll=yes:)

This tutorial illustrates the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client. The latest version of the transcoding library, as of 1 August 2013 is: '''sng-tc-linux-1.3.4.1.x86_64.tgz'''

This tutorial illustrates the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client. The latest versions of the transcoding library, as of 1 August 2013 is: '''sng-tc-linux-1.3.4.1.x86_64.tgz'''

All functions properly handle retransmissions. More details can be found in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]]. A basic configuration file is shown below:

modparam("usrloc", "db_url", "mysql://root:sangoma@localhost/opensips")

Due to the limitations of the transcoding library, transcoding sessions can only be created if both codec A and codec B are known. Since this cannot be accomplished with a standard SIP call flow, the module ''restricts'' the receiving UA to perform ''late SDP negotiation''. The diagram below explains this better:

!!!! Call flow with the ''sngtc'' OpenSIPS module

Due to the limitations of the transcoding library, transcoding sessions can only be created if both codec A and codec B are known. Since this cannot be accomplished with a standard SIP call flow, the module ''restricts'' the receiving UA to perform ''late SDP negotiation''.

----
!!!! Using the ''sngtc'' module

http://www.opensips.org/images/tc-setup.png

http://www.opensips.org/images/dialog-establish-tc.png

http://www.opensips.org/images/tc-setup.png

http://www.opensips.org/images/tc-setup.jpg

[[Attach:tc-setup.png]]

As mentioned in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]], transcoding may only be done if the UAC performs early SDP negotiation.

setsflag(FAIL_TRANS_FLAG);

} else if (is_method("ACK")) {

}

if (is_method("INVITE"))
sngtc_offer();

sngtc_callee_answer("11.12.13.14", "11.12.13.14");

!!!!! Documentation -> [[Documentation.Tutorials | Tutorials ]] -> Voice Transcoding in OpenSIPS using Sangoma D-series Cards

# sngtc_offer - called at any INVITE request (re-INVITES too), deletes the INVITE SDP body
# sngtc_callee_answer - called at 200 OK responses, intersects codec offers, creates transcoding sessions if necessary
# sngtc_caller_answer - called at ACK requests, adds an SDP body to the ACK request

@]

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

/* uncomment the following lines to enable debugging */
#fork=no
#log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
based on revers DNS on IPs */

####### Modules ########

loadmodule "rr.so"

loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "sngtc.so"

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

route {

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

force_rport();

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing

if (loose_route()) {

if (is_method("BYE")) {

setsflag(FAIL_TRANS_FLAG); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
sngtc_offer();
record_route();

} else if (is_method("ACK"))
sngtc_caller_answer();

# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {

if (is_method("ACK")) {
if (t_check_trans()) {
# non loose-route, but stateful ACK; must be an 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;
}

# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();

# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}

# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();

if (is_method("INVITE"))
sngtc_offer();

# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}

if (is_method("REGISTER"))
{
if (!save("location"))
sl_reply_error();

exit;
}

if ($rU == NULL) {
sl_send_reply("484","Address Incomplete");
exit;
}

# do lookup with method filtering
if (!lookup("location","m")) {

t_newtran();
t_reply("404", "Not Found");
exit;
}

# when routing via usrloc, log the missed calls also
setflag(DB_MISSED_FLAG);
route(1);
}

route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_reply("2");
t_on_failure("1");
}

if (!t_relay()) {
send_reply("500", "Internal Error");
}

exit;
}

onreply_route[2] {

if ($rs == 200)
sngtc_callee_answer("38.126.203.98", "38.126.203.98");
}

failure_route[1] {

if (t_was_cancelled()) {
exit;
}

if (next_branches()) {
t_on_reply("2");
t_on_failure("1");
t_relay();
}
}

All functions properly handle retransmissions. More details can be found in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]]. An example configuration file is shown below:

[@
debug=6
log_stderror=yes
log_facility=LOG_LOCAL0

fork=yes
children=6

memdump=1

auto_aliases=no

listen=udp:eth1:5060

disable_tcp=yes

disable_tls=no

exec_msg_threshold=200000

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

#set module path
mpath = "modules/"

loadmodule "exec.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "maxfwd.so"
loadmodule "sipmsgops.so"
loadmodule "mi_fifo.so"
loadmodule "rr.so"
loadmodule "db_mysql.so"
loadmodule "uri.so"
loadmodule "dialog.so"

modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "disable_6xx_block", 1)
modparam("tm", "onreply_avp_mode", 1)

modparam("rr", "append_fromtag", 0)


modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


modparam("db_mysql", "exec_query_threshold", 200000)


modparam("uri", "use_uri_table", 0)


modparam("dialog", "db_mode", 0)
modparam("dialog", "default_timeout", 3600)
#modparam("dialog", "db_update_period", 5)
modparam("dialog", "profiles_no_value", "inbound ; outbound ; shared")

#modparam("dialog", "profiles_with_value", "caller ; myprofile; share")
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "db_url", "mysql://root:sangoma@localhost/opensips_1_9")
modparam("usrloc", "nat_bflag", "NAT_BFLAG")

loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSIST_FLAG")
#modparam("registrar", "attr_avp", "$avp(attr_avp)")

loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we 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)
modparam("acc", "failed_transaction_flag", "FAIL_TRANS_FLAG")
/* account triggers (flags) */
modparam("acc", "db_flag", "DB_CDR_FLAG")
#modparam("acc", "cdr_flag", "DB_CDR_FLAG")
modparam("acc", "db_missed_flag", "DB_MISSED_FLAG")
modparam("acc", "multi_leg_info","leg_id=$avp(leg_id)")
modparam("acc", "multi_leg_bye_info","call_cost=$avp(call_cost);used_min=$avp(used_min)")

loadmodule "cfgutils.so"
loadmodule "nathelper.so"


modparam("registrar","received_avp", "$avp(42)")
modparam("nathelper","received_avp", "$avp(42)")

modparam("cfgutils", "lock_pool_size", 64)
@]

All functions properly handle retransmissions. More details can be found in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]]. An example configuration file is presented below:

# sngtc_offer - called at any INVITE request (re-INVITES too)
# sngtc_callee_answer - called at 200 OK responses
# sngtc_caller_answer - called at ACK requests

- sngtc_offer - called at any INVITE request (re-INVITES too)
- sngtc_callee_answer - called at 200 OK responses
- sngtc_caller_answer - called at ACK requests

The '''sngtc module''' exports 3 functions. They are called upon receiving INVITE, 200 OK and ACK messages. A short overview on their purpose would be:
* sngtc_offer - called at any INVITE request (re-INVITES too)
* sngtc_callee_answer - called at 200 OK responses
* sngtc_caller_answer - called at ACK requests

More details can be found in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]]. An example configuration file is presented below:

The necessary firmware for the D-series transcoding cards can be set up using the ''sngtc_tool''. The cards which require PCI connectivity (D100 and D500) also need additional kernel drivers. Please refer to the [[http://wiki.sangoma.com/Sangoma-Media-Transcoding-Product-Selection|Sangoma wiki]] for installation tutorials.

The necessary firmware for the D-series transcoding cards can be set up using the ''sngtc_tool''. The cards which require PCI connectivity (D100 and D500) require the installation of additional kernel drivers. Please refer to the [[http://wiki.sangoma.com/Sangoma-Media-Transcoding-Product-Selection|Sangoma wiki]] for installation tutorials.

All D-series trascoding cards require the proper firmware to be set up with the ''sngtc_tool''. The cards which require PCI connectivity (D100 and D500) require the installation of additional kernel drivers. Please refer to the [[http://wiki.sangoma.com/Sangoma-Media-Transcoding-Product-Selection|Sangoma wiki]] for installation tutorials.

By default it logs to /var/log/sngtc_server.log

By default it logs to ''/var/log/sngtc_server.log''

$ /etc/init.d/sngtc_server_ctrl start

This tutorial illustrates the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client.

!!!!! Documentation -> [[Documentation.Tutorials | Tutorials ]] -> Voice Transcoding using Sangoma's D-series Cards

The '''sngtc module''' exports 3 functions. They are called upon receiving INVITE, 200 OK and ACK messages. More details can be found in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]].

Since the D-series transcoding cards are connected through either a PCI slot or simply an Ethernet connector, they cannot be assigned global IPs. Consequently, the module will write the local, private IP of the card in the SDP answers sent to each of the endpoints. Since this will not work with non-local UAs, the optional parameters force the RTP listen interface for each UA. This way, the script writer can enforce a global IP for the incoming RTP (which can be port forwarded to a transcoding card).

@]

The transcoding sessions on the Sangoma cards are closed in a transparent manner, based on dialog callbacks.

The '''sngtc module''' exports 3 functions. They are called upon receiving INVITE, 200 OK and ACK messages.

...

...

When receiving INVITE requests:

Handling 200 OK responses:

}
@]

\\

And ACK requests:

[@
route {
...
if (has_totag()) {
if (is_method("ACK"))
sngtc_caller_answer();
}
}
...

}
@]

\\

At 200 OK time:

[@
onreply_route[1] {
if ($rs == 200)
sngtc_callee_answer("38.126.203.98", "38.126.203.98");

The '''sngtc''' module exports 3 functions. They are called upon receiving INVITE, 200 OK and ACK messages.

\\

At INVITE time:

[@
route {
...
if (is_method("INVITE"))
sngtc_offer();
...
}
@]

Installing the ''sngtc_server'' is straightforward and also documented in the [[http://wiki.sangoma.com/FreeSWITCH-D100-Single-Server-Installation|Sangoma wiki]]. After doing the configuration, it can be started using the init script:

!!!! Using the ''sngtc'' OpenSIPS module

The '''sngtc''' module exports 3 functions. They are called upon receiving INVITE, 200 OK and ACK messages.

$ /etc/init.d/sngtc_server_ctl start

Installing the ''sngtc_server'' is straightforward and also documented in the [[http://wiki.sangoma.com/FreeSWITCH-D100-Single-Server-Installation|Sangoma wiki]]. After doing the configuration, you can start it with the init script:
[@
/etc/init.d/sngtc_server_ctl start
@]

Installing the ''sngtc_server'' is straightforward and also documented in the [[http://wiki.sangoma.com/FreeSWITCH-D100-Single-Server-Installation|Transcoding wiki page]].

This tutorial will illustrate the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client.

!!!! Installing the transcoding card

The D-series transcoding cards which require PCI connectivity (D100 and D500) require the installation of additional kernel drivers. Please refer to the [[http://wiki.sangoma.com/Sangoma-Media-Transcoding-Product-Selection|Sangoma wiki]] for installation tutorials.

----

!!!! Using the ''sngtc'' OpenSIPS module

----
!!!! Setting up the ''sngtc_server''

----
!!!! Using the ''sngtc'' OpenSIPS module

!!!!! Documentation -> [[Documentation.Tutorials | Tutorials ]] -> Performing Voice Transcoding using Sangoma's D-series Cards
This page has been visited {$PageCount} times.
(:toc-float Table of Content:)
----

!!!! Tutorial Overview

This tutorial will illustrate the required steps in order to perform audio transcoding with the D-series cards manufactured by Sangoma, using the ''sngtc_server'' daemon and the OpenSIPS ''sngtc'' module as its client.

\\

As stated in the [[http://www.opensips.org/html/docs/modules/devel/uri.html|module documentation]], transcoding may only be done if the UAC performs early SDP negotiation.

The solution below requires no changes at all on the OpenSIPS side ( because it relies on a WebSocket to SIP gateway ), thus it can be easily integrated with '''0''' side-effects to your existing deployment.
Secondly, the solution uses entirely open source components ( both for the server and the client side ) - which gives you great flexibility on the server side, as well as on the client side - since you can easily integrate the client into your existing client portals.

!!!! Platform Components

* We will be using OpenSIPS as the core part of the platform, handling all the SIP traffic. For a basic tutorial of setting up OpenSIPS see the [[ http://www.opensips.org/Documentation/Tutorials-GettingStarted | getting started tutorial ]]
* [[ http://oversip.net | OverSIP]] will be used as a WebSocket to SIP gateway - all the websocket traffic will reach OverSIP, which will decapsulate the SIP traffic and relay it to the OpenSIPS proxy. OverSIP packages are available for Debian based systems and detailed installation instructions are available for other operating systems as well - see OverSIP deployment tutorial [[ http://oversip.net/documentation/1.3.x/installation/ | here ]]

* For the SIP client for testing purposes, we will be using [[ http://sipml5.org/ | sipML5 ]] . It can be downloaded from [[http://code.google.com/p/sipml5/source/checkout | here]] and installation is very straight-forward : just copy the sources to your /var/www/ folder. The client's media stack relies on WebRTC and the client can be used to connect to any SIP or IMS network from your preferred browser to make and receive audio/video calls and instant messages. Furthermore, sipML5 should work on any web browser supporting WebRTC but we highly recommend using [[ https://www.google.com/intl/en/chrome/browser/ | Google Chrome ]] or [[http://nightly.mozilla.org/ | Firefox Nightly]] for testing.

----