Table of Contents
List of Examples
The Sangoma transcoding module offers the possibility of performing voice transcoding with the D-series transcoding cards manufactured by Sangoma. The module makes use of the Sangoma Transcoding API in order to manage transcoding sessions on the dedicated equipment. For the cards in the network to be detected, the Sangoma SOAP server must be up and running (sngtc_server daemon).
The module performs several modifications in the SDP body of SIP INVITE, 200 OK and ACK messages. In all transcoding scenarios, the UAC performs early SDP negotiation, while the UAS does late negotiation. This way, OpenSIPS becomes responsible for intersecting the codec offer and answer, together with the management of transcoding sessions on the Sangoma cards.
This scenario brings about a couple of restrictions:
UACs MUST only perform early SDP negotiation
UASs MUST support late SDP negotiation (rfc 3261 requirement)
Since the sngtc_node library performs several memory allocations with each newly created transcoding session, the module uses a dedicated process, responsible for the management of the above-mentioned sessions. The sangoma_worker process communicates with the OpenSIPS UDP receivers through a series of pipes.
The following libraries or applications must be installed before running OpenSIPS with this module loaded:
sngtc_node library - download from Sangoma, compile and install .
sngtc_server up and running.
The function strips off the SDP offer from a SIP INVITE, thus asking for another SDP offer from the opposite endpoint (late negotiation).
The following error codes may be returned:
-1 - SDP parsing error
-3 - internal error / no more memory
The function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
Example 1.1. sngtc_offer usage
...
if (is_method("INVITE")) {
t_newtran();
create_dialog();
sngtc_offer();
}
...
Handles the SDP offer from 200 OK responses, intersects both offers with the capabilities of the transcoding card and creates a new transcoding session on the card only if necessary. It then rewrites the 200 OK SDP so that it contains the information resulted from the codec intersection.
Parameters explained:
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).
listen_if_A - the interface where the UAC (the caller) will send RTP after the call is established (IP from the 'c=' SDP line(s))
listen_if_B - the interface where the UAS (the callee) will send RTP after the call is established (IP from the 'c=' SDP line(s))
The following error codes may be returned:
-1 - SDP parsing error
-2 - failed to create transcoding session
-3 - internal error / no more memory
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
Example 1.2. sngtc_callee_answer usage
...
onreply_route[1] {
if ($rs == 200)
sngtc_callee_answer("11.12.13.14", "11.12.13.14");
}
...
Attaches an SDP body to the caller's ACK request, so that it matches the late SDP negotiation done by the UAS.
The following error codes may be returned:
-3 - internal error / no more memory
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
Example 1.3. sngtc_caller_answer usage
...
if (has_totag()) {
if (loose_route()) {
...
if (is_method("ACK"))
sngtc_caller_answer();
}
...
}
...