SBC module

Copyright (C) 2010-2011 Stefan Sayer

The SBC application is a highly flexible high-performance Back-to-Back
User Agent (B2BUA). It can be employed for a variety of uses, for example 
topology hiding, NAT handling, From/To modification, enforcing SIP Session Timers, 
identity change, SIP authentication, RTP relaying, transcoding, accounting,
registration cache, RTP bandwidth limits.

 o B2BUA with topo hiding or transparent modes
 o flexible call profile based configuration
 o online reload of call profiles
 o From, To, RURI, Contact, Call-ID update
 o Registration caching
 o SIP NAT handling
 o RTP bridging
 o Header and message filter
 o adding arbitrary headers
 o reply code translation
 o SIP authentication
 o SIP Session Timers
 o Fixing Call Transfers (Replaces in REFER target and INVITE)
 o call timer
 o prepaid accounting
 o CDR generation
 o call teardown from external control through RPC
 o transcoding
 o ...

SBC Profiles
All features are set in an SBC profile, which is configured in a
separate configuration file with the extension .sbcprofile.conf. Several
SBC profiles may be loaded at startup (load_profiles), and can be
selected with the active_profile configuration option. The
active_profile option is a comma-separated list, the first profile that
matches, i.e. is non-empty, will be used.

In this list a profile may be selected

 o statically (active_profile=<profile_name>)

 o depending on user part of INVITE Request URI (active_profile=$(ruri.user))

 o depending on "profile" option in P-App-Param header

 o depending on a Request URI parameter (active_profile=$rP(param_name))

 o using any replacement pattern (see below), especially regex maps $M(val=>map)

By using the latter options, the SBC profile for the call can also be
selected in the proxy.


  In order to have all calls coming from source IP 10.0.* going to
  'internal1' profile, all calls coming from source IP 10.1.* going to
  profile, then for calls coming from other IP addresses those to RURI-domain go to 'iptel' profile, and all other calls being refused, we
  could set

  ~~~~~~~~~ sbc.conf ~~~~~~~~~
  ~~~~~~~~~~~~~~~~~~ ~~~~~~~~~

  ~~~~~~~~~ src_ipmap.conf ~~~

  ~~~~~~~~~ rurimap.conf ~~~~~>iptel

SBC profile reload
The SBC profiles may be reloaded while the server is running. A set of
(python) scripts is provided and installed to trigger the reload
(through XMLRPC): 

  sems-sbc-list-profiles                        list loaded profiles
  sems-sbc-reload-profile  <name>               reload a profile (from its
                                                .conf file) 
  sems-sbc-reload-profiles                      reload all profiles (from
                                                .conf files)
  sems-sbc-load-profile <name> <conf_file>      load a profile from a file
                                                (e.g. new profile or file
                                                path changed)
  sems-sbc-get-activeprofile                    get active_profile
  sems-sbc-set-activeprofile <active_profile>   set active_profile
  sems-sbc-teardown-call <call_ltag>            tear down call (use e.g.
                                                sems-list-active-calls to
                                                get the ltag)

The xmlrpc2di module must be loaded and the XMLRPC control server bound
to port 8090 for the scripts to work.

For tracking file revisions and changes, the MD5 hash sum is printed on
profile load and reload, and returned as information by the scripts and
the DI management commands. An MD5 hash to compare checksums of profile
files can also be generated with the md5sum(1) tool. 

Alternatively, the reload functions can be accessed by json-rpc v2 if
the jsonrpc module is loaded. The expected parameters to all functions
are in a dictionary with 
   'name' :          profile name
   'path' :          profile conf file path
   'active_profile': active profile (string)
Return code is [200, "OK", <result dictionary>] on success, or 
[<error code>, <error reason>] on failure.

Replacement patterns - active_profile, RURI, From, To, Contact, etc
In SBC profile the appearance of the outgoing INVITE request can be set,
by setting RURI, From and To parameters. If any of those parameters is not
set, the corresponding value of the incoming request is used.

The values that are set can contain patterns, which are set to values taken
from the incoming INVITE request. The syntax loosely follows sip-router's
pseudo variables. Any of the RURI, From and To values can contain any elements,
e.g. the request-URI can be set to the user part of the P-Asserted-Identity
header combined with the host part of the To.

The patterns which can be used are the following:

  $r (or $r. if something follows) - R-URI
  $f (or $f. if something follows) - From
  $t (or $t. if something follows) - To
  $a (or $a. if something follows) - P-Asserted-Identity
  $p (or $p. if something follows) - P-Preferref-Identity

  $fu  - From URI
  $fU  - From User
  $fd  - From domain (host:port)
  $fh  - From host
  $fp  - From port
  $fH  - From headers
  $fP  - From Params
  $fP(param) - From Param 'param'
  $ft  - From tag
  $fn  - From display name

  $tu  - To URI
  $fU  - To User

  $ru  - R-URI URI
  $rU  - R-URI User

  $ai  - P-Asserted-Identity URI (alias to $au)
  $au  - P-Asserted-Identity URI
  $aU  - P-Asserted-Identity User

  $pi  - P-Preferred-Identity URI (alias to $pu)
  $pu  - P-Preferred-Identity URI
  $pU  - P-Preferred-Identity User

  $ci  - Call-ID

  $si  - source (remote) IP address
  $sp  - source (remote) port

  $Ri  - destination (local/received) IP address
  $Rp  - destination (local/received) port
  $Rf  - local/received interface id (0=default)
  $Rn  - local/received interface name
         ('default', 'intern', ... as set in sems.conf)
  $RI  - local/received interface public IP (as set in sems.conf)

  $P(paramname) - paramname from P-App-Param
      P-App-Param: u=myuser;p=mypwd;d=mydomain

  $H(headername) - value of header <headername>
    o P-Caller-Uuid: 0004152379B8

    o P-NextHop:

  $HU(headername) - header <headername> (as URI) User
  $Hd(headername) - header <headername> (as URI) domain (host:port)

    o P-SomeNH-URI: sip:user@

  $M(value=>regexmap) - map a value (any pattern) to a regexmap (see below)
    Example: $M($fU=>usermap)

  $_*(value) - string modifiers: 
   $_u(value)   - value to uppercase (e.g.: $_u($fh) From host in uppercase)
   $_l(value)   - value to lowercase (e.g.: $_l($fh) From host in lowercase)
   $_s(value)   - length of value (e.g.: $_s($fU) string length of From user)
   $_5(value)   - MD5 of value

  \\  -> \
  \$  -> $
  \*  -> *
  \r  -> cr  (e.g. use \r\n to separate different headers in append_headers)
  \n  -> lf
  \t  -> tab

If a quotation mark (") is used, it needs to be escaped with a backslash in
the sbc profile configuration file.
   From="\"Anonymous\" <sip:anonymous@invalid>"

If a space is contained, use quotation at the beginning and end.
   To="\"someone\" <$>"

Regex mappings ($M(key=>map))

A regex mapping is a (sorted) list of "regular expression" => "string
value" pairs. The regex mapping is executed with a key - any string,
replacement pattern or combination - and the first regular expression
that matches returns the "string value". 

Regex mappings are read from a text file, where each line corresponds to one
regex=>value pair. The mappings to load on startup are set with the regex_maps
config option, the file name from where it is loaded is "<mapping
name>.conf" in the plugin config path.

Mappings can also loaded into the running server by using the
setRegexMap DI function or the included sems-sbc-*-regex-* scripts:

  sems-sbc-set-regex-map <name> <file>      load a regex map from a file
  sems-sbc-get-regex-map-names              list regex map names

 Example regex map:
   ~~~~~~~ usermap.conf ~~~~~~
   # this is a comment

Setting Call-ID
For debugging purposes, the call-id of the outgoing leg can be set to depend on
the call-id the first leg, by setting the Call-ID parameter.


  If the incoming call leg had "Call-ID: 3c2d4b9a6b6f-hb22s7k9n0iv", the
  outgoing leg will have "Call-ID: 3c2d4b9a6b6f-hb22s7k9n0iv_leg2".

If Call-ID is not set, a standard unique ID is generated by SEMS, of the form

Outbound proxy and next hop

An outbound proxy may be set with the outbound_proxy option. If this is
not set, the outbound_proxy option of sems.conf is used, if that one is set.
Setting an outbound proxy will add a route header.

force_outbound_proxy forces the outbound proxy as first route and request URI
also for in-dialog requests. Note that this is NOT RFC3261 compliant (section
2.2 Requests within a Dialog, 12.2.1 UAC Behavior).

The next hop (destination IP[:port] of outgoing requests) can be set
with the next_hop option. next_hop port defaults to 5060 if not set or
empty. Multiple alternatives for the next hop can be set, also with their
respective transports, e.g.

Usually, replies are sent back to where the request came from
(honoring rport), but if next_hop should be used nevertheless,
next_hop_for_replies profile option can be set to "yes".

patch_ruri_next_hop=yes sets the option to overwrite RURI in the
outgoing request with the selected next hop (whether set through
next_hop or determined otherwise).

These settings apply only for the UAC side, i.e. the outgoing side of
the initial INVITE.

Registration caching
If registration caching is activated, SEMS SBC saves the contact of REGISTER
messages in a local registration cache (in-memory) along with information of
where the registration was received from and over which transport, interface
etc. SEMS SBC then replaces the contact with a locally generated alias that
points to itself, and sends it to the upstream registrar. When a message comes
from the registrar, SEMS SBC looks up the alias, retargets it from the
registration cache (sets RURI), and sends it over the saved transport to the
proper remote address where the client is (e.g. behind NAT) by setting next_hop
etc properly.

To activate the registration cache, use the option
(for both REGISTER and other messages). With the parameters
 min_reg_expires and max_ua_expires
it can be controlled how long the registration to the upstream registrar should
persist and how short it should be to the UA. E.g. if the UA should periodicly
re-REGISTER every 60 seconds, but to the upstream registrar the registration should
persist 1h, min_reg_expires=3600 and max_ua_expires=60 should be set.

For a local registrar (i.e. operation without an upstream registrar), see the 'registrar'
call control module.

Headers and messages may be filtered. A filter can be set to 
 o transparent - no filtering done

 o whitelist - only let items pass that are in the filter list
 o blacklist - filter out items that are in the filter list

Note that ACK messages should not be filtered.

Codec filter
The SDP body of INVITE/200, UPDATE/200 and ACK may be filtered for codecs
with the sdp_filter and sdpfilter_list call profile options. If sdp_filter is 
set to transparent, the SDP is parsed and reconstructed (SDP sanity check).
Codecs may be filtered out by their payload names in whitelist or blacklist
modes. The payload names in the list are case-insensitive (PCMU==pcmu).

The s, u and o-lines of the SDP can be anonymized with the setting

Codec preference
Payloads within SDP body might be reordered by SBC so clients might be forced to
use prioritzed codecs. 

The priority is given in each call leg independently via codec_preference and
codec_preference_aleg call profile options. These options contain list of codecs
ordered by priorities separated by coma. Payload names are case-insensitive,
clock rate is considered if given.

Codecs are ordered when filtering is done, for SDP offer and for SDP answer as
well. Ordering is applied on each media stream in the SDP. Codecs present in SDP
which are not listed in codec_preference resp. codec_preference_aleg preserve
their order.

for example:

  incoming SDP:

        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:97 speex/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:103 G726-24/8000
        a=rtpmap:104 G726-32/8000
        a=rtpmap:105 G726-40/8000
        a=rtpmap:98 speex/16000
        a=rtpmap:99 speex/32000
        a=rtpmap:102 G726-16/8000
        a=rtpmap:101 telephone-event/8000

  outgoing SDP:
        a=rtpmap:97 speex/8000
        a=rtpmap:98 speex/16000
        a=rtpmap:99 speex/32000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:103 G726-24/8000
        a=rtpmap:104 G726-32/8000
        a=rtpmap:105 G726-40/8000
        a=rtpmap:102 G726-16/8000
        a=rtpmap:101 telephone-event/8000

Call profile options for choosing codec preferences:


    List of codec preferences describing how to reorder codecs in SDP sent from
    caller to callee.

    List of codec preferences describing how to reorder codecs in SDP sent from
    callee to caller.

SIP NAT handling
dlg_nat_handling=yes makes SEMS SBC learn the proper remote address (port, transport,
...) from the received message and use that in following in-dialog requests. Enable
this option when handling far end NATs.

RTP relay
RTP can be bridged through the SBC. Where without rtprelay, A call would go only
with the signaling through the SBC, in rtprelay mode, the connection address in
SDP messages will be replaced to the one of SEMS, such that caller and callee
send RTP media to SEMS. SEMS then relays the RTP packets between the two sides.

RTP relay can be enabled by setting

The SBC detects if UAs indicate that they are behind NAT by setting
a=direction:active in SDP, and goes into passive mode until it receives
the first packet from the NATed client, from which it learns the remote
address. This mechanism is called "symmetric RTP".

Symmetric RTP (starting in passive mode) can also be forced by setting
the rtprelay_force_symmetric_rtp=yes sbc profile option. Symmetric RTP
is enabled if rtprelay_force_symmetric_rtp evaluates to anything other
than "" (empty string), "0" or "no".

Some ser/sip-router/kamailio/*ser configurations add flag 2 in a header
P-MsgFlags header to the INVITE to indicate forcing of symmetric
RTP. With the sbc profileoption


the SBC honors this and sets symmetric RTP accordingly.

With the option rtprelay_dtmf_filtering=yes the SBC filters out RTP DTMF
(RFC2833 / RFC4733) packets in relayed streams.

If rtprelay_dtmf_detection=yes is set, DTMF from RTP packets is detected
and can be used to control applications, e.g. special call flows, implemented with
the extended call control API. Note that the call needs to be added to
the media processor in order for DTMF events to be processed.

The SBC is able to do transcoding together with relaying. 

To trigger transcoding you have to configure transcoder_codecs to a set
of codecs which are understand by SEMS and enable transcoder via
enable_transcoder call profile option.

Transcoder codecs are appended to the end of outgoing SDP (advertising that
their priority is lower than priority of original codecs) and allow to the other
party to choose from more codecs than used in the original SDP. 

  for example:

    Caller understands only PCMA codec, callee: understands only PCMU codec.

    incoming SDP offer: PCMA
    outgoing SDP offer: PCMA, PCMU
    incoming SDP answer: PCMU
    outgoing SDP answer: PCMU, PCMA

    Caller generates RTP with PCMA payload, SEMS in between transcodes to PCMU
    and sends PCMU RTP to the callee.

    Callee generates RTP with PCMU payload, SEMS in between transcodes to PCMA
    and sends PCMA RTP to the caller.

In case of another codec preference you can configure codec_preference resp.
codec_preference_aleg as described above and choose if codecs should be ordered
before adding transcoder codecs or after using prefer_existing_codecs resp.
prefer_existing_codecs_aleg call profile options.

Transcoding related call profile options:


    Possible values: always, on_missing_compatible, never

    The value "on_missing_compatible" causes that transcoder is enabled only in
    case it seems to be necessary. See the option callee_codeccaps below.


    Callee's codec capabilities used when enable_transcoder is set to
    "on_missing_compatible". In this case the set of codecs in input SDP of
    initial INVITE is compared to the list of codecs given by this option. If
    there is no match, transcoder is activated.


    List of codecs which can be transcoded.


    Describes if codecs present in SDP from caller are preferred to added
    transcoder codecs.

    If this parameter is set to "yes" transcoder codecs are added at the end of
    codec lists in SDP body AFTER ordering using codec_preference is done.

    If this parameter is set to something else, transcoder codecs are
    added BEFORE ordering using codec_preference is done and thus may
    become preferred ones. 

    Describes if codecs present in SDP from callee are preferred to added
    transcoder codecs.

    If this parameter is set to "yes" transcoder codecs are added at the end of
    codec lists in SDP body AFTER ordering using codec_preference_aleg is done.

    If this parameter is set to something else, transcoder codecs are
    added BEFORE ordering using codec_preference_aleg is done and thus
    may become preferred ones. 

Transcoder statistics can be checked via "printCallStats" SBC DI method or can
be put into additional headers within reply generated to OPTIONS request. To
achive that set global options: options_transcoder_out_stats_hdr and

  With this settings:


  the OPTIONS reply will contain these headers with statistics:

    P-Transcoder-Stats-In: pcma=1,pcmu=0,speex=1
    P-Transcoder-Stats-Out: pcma=1,pcmu=0,speex=1

The statistics are separated for codecs used for reading (data are read using
that codecs) and for codecs used for writing (data are sent using that codecs).
Note that relaying can be active in one direction and transcoding in the other
so the in/out (read/write) numbers need not to match.

 - currently only audio streams are relayed through or transcoded
 - usage of transparent vs. non-transparent SSRC and sequence numbers is a bit
   tricky when transcoding is possible and let on correct user configuration for
   now (sometimes one of the variants is more suitable, unless we will be
   propagating SSRC changes in incoming RTP neither of them will be working for
 - handling of "on hold" streams when transcoding is in use can cause RTP media
   send in hold state (sendonly stream) though they need not to be sent (caused
   by buffering strategy)
 - the statistics are just approximation of real situation because multiple
   codecs can be used in one stream at the same time but only one (the last used
   one) is reported
Adding headers
Additional headers can be added to the outgoing initial INVITE by using the
append_headers call profile option. Here, several headers can be separated with
\r\n. All replacement patterns from above can be used.

 append_headers="P-Received-IP: $Ri\r\nP-Received-Port: $Rp"
 append_headers="P-Source-IP: $si\r\nP-Source-Port: $sp\r\n"
 append_headers="P-Original-URI: $r"

Response code translations
Response codes and reasons may be translated, e.g. if some 6xx class
replies need to be changed to 4xx class replies.

 reply_translations="603=>488 Not acceptable here"

Here, all 603 replies received on one leg will be sent out as 488 reply with
the reason string "Not acceptable here".

Entries are separated in the reply_translations list with a pipe symbol (|).

 reply_translations="603=>488 Not acceptable here|600=>406 Not acceptable"

Warning: Changing response codes, especially between different response
         code classes, can seriously mess up everything. Use with caution
         and only if you know what you are doing!

Fixing Call Transfers (Replaces in REFER target and INVITE)
Using the profile options fix_replaces_inv and fix_replaces_ref Replaces
can be fixed for call transfers going through the SBC.


For situations where the call transfer is handled by the UAs (phone handles
the REFER), the Replaces should be fixed in the INVITE message (fix_replaces_inv=yes).

For situations where a PBX handles the call transfer (handles the REFER),
the Replaces should be fixed in the REFER message (fix_replaces_ref=yes).

Reliable 1xx (PRACK)

Reliable 1xx (PRACK) extension (3262) is supported in a transparent mode,
i.e. the RSeq header is relayed and RAck CSeq is translated properly.

SIP authentication
The SBC can perform SIP digest authentication. To use SIP authentication, the
uac_auth module needs to be loaded.

SIP authentication is enabled by the following parameters, separately for both
call legs:

# Authentication for B leg (second/callee leg):
   enable_auth       "yes" or "no"
   auth_user         authentication user
   auth_pwd          authentication password
# Authentication for A leg (first/caller leg):
   enable_aleg_auth  "yes" or "no"
   auth_aleg_user    authentication user
   auth_aleg_pwd     authentication password

Note: The 'A' leg is always the first leg, the one from the caller. 'B' leg is
the one to callee:
 caller <--- A (first) leg ---> SEMS <--- B (second) leg ---> callee


NOTIFY pass-through
If in-dialog NOTIFY messages which do not belong to an established subscription
should not be passed through, the profile option 
may be set. This defaults to yes.

SIP Session Timer configuration
If SIP Session Timers are enabled for a profile, the session timers values
(session_refresh, minimum_timer etc) can be configured either in sbc.conf
or in the profile configuration, which overrides the sbc.conf configuration.

SIP Session Timers may be configured for each leg individually.
enable_session_timer overrides enable_aleg_session_timer if that one is
not set: SST may be disabled on the A (caller) leg by setting
enable_aleg_session_timer=no.  If enable_session_timer=yes and
enable_aleg_session_timer not set, SST is enabled for both
legs. Likewise, if aleg_session_expires etc. is not set, the SST
configuration of the B leg is used (session_expires, minimum_timer etc).

Call hold configuration

SBC detects hold offer in SDP and according to configured parameters it can
alter the hold requests passing through. (this may be handy for example if there
is need to change "sendonly" to "sendrecv" for correct passing hold music
through NATs)

hold_alter_b2b_aleg / hold_alter_b2b_bleg

  If set to "yes" SBC alters B2B hold requests according to hold settings.

  If set to "no" hold settings is used for locally generated hold requests only.

hold_zero_connection_aleg / hold_zero_connection_bleg

  If set to "yes" all connections within SDP are replaced with

  If set to "no" SBC tries to put its own media IP everywhere (to replace Note that if SBC is not doing rtp relay, it can not replace IP
  with its own address and the original one is kept there.

hold_activity_aleg / hold_activity_bleg

  Force "stream activity" (send/recv) on hold offer.

  Possible values: sendrecv, sendonly, recvonly, inactive

Call control modules
Call control (CC) modules for the sbc application implement business
logic which controls how the SBC operates. For example, a CC module can
implement concurrent call limits, call limits per user, enforce other
policies, or implement routing logic. 

Call control (CC) modules should be loaded using the load_cc_plugins
option in sbc.conf, or loaded later into the server by the
sems-sbc-loadcallcontrol-modules script (loadCallcontrolModules DI function).

Multiple CC modules may be applied for one call. The data that the CC
modules get from the call may be freely configured. Call control modules
may also be applied through message parts (replacement patterns).

  Limiting From-User to 5 parallel calls, and 90 seconds maximum call duration:

  Applying 90 seconds maximum call duration and other call control from
  a header:

   SIP message:
    INVITE SIP/2.0
    P-CallControl: cc_pcalls;uuid=$rU, cc_pcalls;uuid=a_user

See also Readme.sbc_call_control.txt.

Call control: Prepaid
Prepaid accounting can be enabled with using a prepaid call control module.
The credit of an account is fetched when the initial INVITE is processed,
and a timer is set once the call is connected. When the call ends, the credit
is subtracted from the user.

For accounting, a separate module is used. This allows to plug several types
of accounting modules.


  Here the account UUID is taken from the P-Caller-Uuid header, and the
  accounting destination from the P-Acc-Dest header.

Credit amounts are expected to be calculated in seconds. The timestamps
are presented in unix timestamp value (seconds since epoch). start_ts is
the initial INVITE timestamp, connect_ts the connect (200 OK) timestamp,
end_ts the BYE timestamp.

The cc_prepaid and cc_prepaid_xmlrpc modules may be used for accounting
modules, or as starting points for integration into custom billing systems.

Call control: Parallel calls limit
Parallel call limits can be enforced by using the parallel calls call
control module.

 Example (limit From-User to max 5 calls):

Call control: Call Timer
A maximum call duration timer can be set with the call timer call
control module. 

 Example (timer taken from P-Timer header):

 Example (maximum 90 seconds):

CDR generation
CDR generation can be enabled with loading a CDR call control module.

The cc_syslog_cdr module writes CDRs to syslog(3) to be processed
by standard syslog utils, e.g. syslog-ng.


See also cc_syslog_cdr module documentation.

Refusing calls

In some configurations, if may be necessary to refuse calls with a
certain error response code and reason. If the refuse_with call profile
option is set, the call is refused with the code and reason
specified. In this case, all other call profile options are ignored,
only the append_headers option has effect.

 refuse_with="403 Invalid Domain $rd"

 refuse_with="606 Not Acceptable"
 append_headers="P-Original-URI: $r\r\nP-Original-To: $t"

Example profiles
 transparent   - completely transparent B2BUA (contains all options in comments)
 auth_b2b      - identity change and SIP authentication (obsoletes auth_b2b app)
 sst_b2b       - B2BUA with SIP Session Timers (obsoletes sst_b2b app)
 call_timer    - call timer (obsoletes call_timer app)
 prepaid       - prepaid accounting (obsoletes sw_prepaid_sip app)
 codecfilter   - let only some low bitrate codecs pass
 replytranslate - swap 603 and 488 response code in replies
 refuse        - refuse all calls with 403 Forbidden
 symmetricrtp  - RTP relay with symmetric RTP for NAT handling

For SIP authentication: uac_auth module
For SIP Session Timers and call timers: session_timer module

x header filter (whitelist or blacklist)
x message filter (whitelist or blacklist)
x SDP filter (reconstructed SDP)
x remote URI update (host / user / host/user)
x From update (displayname / host / host/user)
x To update (displayname / host / host/user)
x SIP authentication
x session timers
x maximum call duration timer
- accounting (MySQL DB, cassandra DB)
x RTP forwarding mode (bridging)
x RTP transcoding mode (bridging)
- overload handling (parallel call to target thresholds)
- call distribution
- select profile on monitoring in-mem DB record
x fallback profile
x add headers
x bridging between interfaces
- rel1xx in non-transparent mode