FOIP lines and devices Flashcards
Describe FOIP lines
FaxMaker FoIP providers use two protocols for use with FaxMaker and FoIP: SIP and H.323. We will cover both protocols and their associated signaling.
Two considerations before reviewing SIP and H.323 is the transport protocol being used as well as the RTP codecs.
Describe FOIP transfer protocol
G.711 and T.38 are transport protocols that can be used with both SIP and H.323. The article Comparing T.38 and G.711 gives a good description of each transport.
When troubleshooting with wireshark the T.30 protocol signaling will not be seen in a G.711 wireshark capture as the information is transmitted via RTP (Real-Time Transport Protocol) streams (audio) versus standard signaling as in T.38. From a support perspective that is the only difference.
Describe RTP codecs
Since G.711 uses RTP streams for the T.30 data, we also need to review the codecs used. These are generally classified as being used in different geographical areas, like E-1 and T-1 with ISDN PRI, but the reasoning for each is the resolution they give to higher or lower range signals.
PCMU, also known as G711μ, G711MU or G711 μ-law: Tends to give more resolution to higher range signals and is used in North America and Japan.
PCMA, also known as G711A, or G711 A-law: Provides more quantization levels at lower signal levels and is used in Europe and Australia.
Describe SIP
Session Initiation Protocol: Protocol for signaling and controlling multimedia communication sessions in applications of internet telephony and is primarily used to setup and terminate voice or video calls. The default port for SIP is UDP 5060.
For our purposes the SIP protocol is the medium by which the fax is transmitted across the internet, or through the customer’s network to the PSTN network. A SIP session is set up and then the communication of the T.30 protocol takes place through the SIP session.
List and describe the 9 most common SIP methods
INVITE
Request to initiate a SIP session. INVITE is sent from the calling party to the called party.
ACK
The called party has accepted the call. ACK is sent from the calling party to the called party.
OPTIONS
The calling party is requesting the called party to respond with its capabilities.
BYE
Request to terminate the session. BYE can be sent by the calling of the called party. It is not necessary for the party receiving a BYE to respond with a BYE.
CANCEL
Cancels pending requests.
REGISTER
The calling party wants to register its current location with a registrar server.
SUBSCRIBE
The calling party requests an update regarding the presence information of the called party.
NOTIFY
Conveys the updated status of self to subscribed parties of self.
MESSAGE
Used to send an instant message.
Describe a SIP session.
A SIP session setup is very similar to the TCP protocol’s three way handshake: SYN, SYN/ACK, ACK. With SIP it is INVITE, OK, ACK. Once this has transpired, the session is setup and communication can begin. The session ends with a BYE, OK signaling. Many other things can transpire, but these are the very basics.
This is shown in the Wireshark capture as it clearly displays the SIP signaling outlined in red. We can also see in the INVITE and the OK that these are G.711 packets, so all of the T.30 protocol information is in the RTP streams that are highlighted in yellow.
In the Wireshark capture below, we can see the G.711 information in red again. Then outlined in blue we see the T.38 packets. It is another invite sequence setting up the T.38 transport mode. Then outlined in purple we can see the start of the T.30 packet information.
In conclusion, the SIP session is setup as the medium for transporting the T.30 protocol from endpoint to endpoint. Once the connection is setup, the T.30 protocol is then transmitted via RTP streams or visibly in the capture.
Describe H.323
H.323 is a recommendation from the ITU Telecommunications Standardization Sector (ITU-T) that defines protocols to provide audio-visual communication sessions on any packet network. It addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely used for voice and video services over IP networks. The default port for H.323 is UDP 1720.
H.323 is not a single protocol but uses a collection of protocols and transport methods. For our purposes H.225 and H.245 are both used. Although it is similar to SIP but requires more bandwidth and is a more complicated series protocols. Both are capable of achieving the goal of FoIP. H.323 is not as commonly used for FoIP as SIP is, but we do see customers who use it occasionally.
H.323 in a capture as not as pretty and easy to follow at a glance as a SIP capture can be. The example below shows some of the common signaling used in an H.323 transmission:
Describe a session with H.323.
The first section, outlined in red is the call setup. This establishes communication between the endpoints and moves to progress the call.
Then next section in blue determines each terminal’s capabilities and sets which terminal is master and which is slave. This is important during a call as the master will determine all disputes, for example if each endpoint attempt to open incompatible media flows, the master will take action to determine the incompatible media flow.
The next step, outlined in green, is where logical channels are opened for communication. Once this takes place an endpoint may then transmit data.
Finally, outlined in purple, the alerting and connect messaging show the connection has been made. The release complete also tears down the call.
Just like with SIP, the H.323 session is the medium to transport the T.30 protocol. Also with SIP, both G.711 or T.38 can be used as a transport. G.711 will be viewed in RTP streams only, while T.38 information will be available for review in the capture.
Describe FOIP devices
Brooktrout, by Dialogic, makes 2 devices. One is a physical card, the other is virtual.
The physical device is a Brooktrout card. These cards have model numbers that depict the features of the card. TR1034+P8V8FH-E1-1N, for example, the 1N on the end shows this card has an Ethernet port to connect to a network for use with FoIP.
The Brooktrout SR140 is a virtual card. It uses the same driver as the physical card and since it is not a physical device, the server can be made virtual.
XCAPI is another virtual device, which allows virtualization of the FaxMaker server. It is made by TE-SYSTEMS.
Describe FOIP logging.
Logging we will fully cover in the Brooktrout and XCAPI courses, but will note the logs used to troubleshoot each device transmissions.
Brooktrout physical card: This will use the ltbtrtxx.txt files where xx = the line number. Brooktrout also has ecc logging which is enabled in the Brooktrout configuration. Wireshark captures cannot be used with this device as the Windows host does not have access to the Ethernet controller on the Brooktrout board.
The Brooktrout SR140 logs will be the ltbrk.txt and the ecc logging described above. A Wireshark capture of the calls is very useful as well, depending on transport protocol being used.
XCAPI, which FaxMaker treats as an ISDN device, produces ISDN logs. Although Wireshark captures can also be used, XCAPI also offers a software suite which includes the ability to capture traces of the calls made on the device. The XCAPI trace can then be sent to support for review.
