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Some aspects concerning IP-telephony (part II)

November 8, 2000



Delay influence in IP/H.323 

Networks with commutation of packets were created for data transfer, and the possibility of voice/fax data transfer in real time mode greatly depends on a delay which is brought in with the signal being transferred. Fig.10 demonstrates a scheme of the VoIP network and the delays occurring there. 

Fig. 10 

Noteworthy is the fact that the delays could cause incorrect functioning of a telephony alarm  in the digital paths E1/T1 on the joint of voice gateways with the equipment of the switched networks. It is because of the fact that H.323 recommendations when appeared were intended for multimedia applications which maintained audio and video conference connection via IP networks. This solution allowed to decrease the cost as compared with systems of a traditional telephony with channel commutation. With the development of IP-telephony it became necessary to connect IP-gateways with Telephony Network of Common Use with E1/T1 digital paths. The gateways connect to digital automatic telephone exchange, using standard telephony alarm Q.931. According to Q.931, time delays between phases of accomplishing alarm commands are set to definite values. However, when interpreting Q.931 commands with H.225/TCÐ/IP, the delays occurring increase the set time between Q.931 commands, and can prevent the protocol functioning. Although the version 2 of H.323 includes H.323v2 Fast Connect, the delays can exceed all the admissible intervals of Q.931 protocol. This makes reasonable organizing of clear channels in IP-telephony networks. 

Clarent Bandwidth Calculator 

Clarent company has developed a program called Clarent Bandwidth Calculator (fig.11) to simplify calculations of a presumable data rate for gateway-gateway IP-channels when transferring voice/fax data. 

Fig. 11 

The results (Kbit/s) are calculated for the local network and for WAN interfaces. The source data are: a codec type, the number of simultaneous conversations, the set value of a voice detector threshold, and reserved bandwidth. In Complex part you can see the results calculated with the usage of information optimal compression technology. In Simplex part there are the figures for a usual gateway for IP-telephony under H.323. Below you can see the results made with the help of the Clarent Bandwidth Calculator for a usual IP-gateway with different combinations of parameters. 

How a WAN channel bandwidth depends on a codec's type and the number of simultaneous conversations

The calculations were implemented for a 30-channel voice gateway working under the control of H.323 gatekeeper via and E1 PRI digital path. The bandwidth of the WAN channel is completely accessible for a telephony traffic and doesn't have a reserve. The trigger level of a voice detector is  30% of the maximum signal amplitude. The network utilize RAS procedure to define the connection of the gateway and the gatekeeper. Figure 12 demonstrates data rates in the WAN channel depending on the number of simultaneous conversations with the following codecs: G.723.1 Low, G.723.1 High, G.729a, NetCoder. The number of I/O calls increases statically. Keep in mind that the data rate is changing constantly in a multichannel IP-gateway. 

Fig. 12 

Here we should notice that:

  • with the static increase of the number of connections the data rate via a gateway-gateway channel has a linear character
  • NetCoder codec (4,8 Kbit/s) requires the higher bandwidth of the channel than the G.723.1 codec (5,3 – 6,3 Kbit/s) does. 
  • NetCoder (8 Kbit/s) and G.729 Annex A codecs have identical speed characteristics, what indicates the similarity of their coding algorithms. 

Switching VAD… 

Fig. 13 illustrates the requirements to the bandwidth of a gateway-gateway channel of H.323 network, depending on a codec's type and a voice detector threshold.

Fig. 13 

The presumable data rate in the IP-channel depending on a voice detector threshold value with the usage of G.711 codec are shown in the figure 14.

Fig. 14 

Facsimile messages transfer via IP

First time ITU-T published a protocol of interconnection of analog facsimile sets in 1980. Those fax machines were reckoned in the Group 3. The protocol consists of several parts which reflect the different stages of  fax transferring. The messages are transferred with the modulated carrier frequency via a usual telephony network. The speed of document image transfer constitutes 64 Kbit/s. 

T.30 specification divides the process of fascimile message transfer  into 5 stages:

A – Dial-up, connection

B – Mutual identification of facsimile sets, speed choice

C – Transfer of a document image

D – Revise of the number of pages

E – Connection break

According to T.4, a document image is transferred in a file in TIFF-F format through modems. At the end of each block you will see EOL (end of line). At the end of the last block it will be written 6 times. 

In 1998 in the second version of H.323 the T.38 specification introduces Fax Relay technology intended for fax transfer in real time mode. 

The practical realization of FoIP is included in all modern IP-gateways of an operator level. However, not all gateways of different manufacturers are compatible when transferring a facsimile traffic. And the realization of T.38  mechanism of different manufacturers turns out the secret information. 

Some more about Fax Relay T.38 you can find on http://www.vocal.com/data_sheets/t38.html. And as for T.38 Fax Relay realization, CISCO Systems company is of great interest. According to CISCO Systems, after connecting to an IP-gateway and transferring the information on the number of the called abonents (Stage A), the facsimile set tries to connect the called fax machine and to set the parameters of a connection speed (Stage B), and the voice gateways on the both ends detect and transmit standard messages of T.30 to the connected fax machines with the connection speed 2,4-14,4 Kbit/s. So, the stages B and D are not necessary to implement. Note, that the process of a document image transfer makes certain requirements to phase distortions of the signal, delays in the path, and to delays when coding/decoding. That's why the most effective for facsimile signal coding are PCM codecs. The process of document image transfer via the packet network is implemented with the usage of UDP protocol and the following technological methods: 

  • Redundant coding and codes with error correction.
  • Repeated transmission of lost or damaged information blocks and packets.
  • Buffering of the received data

Fig.15 shows the results of data rate calculations in IP-channel for different number of simultaneous calls with usage of T.38 Fax Relay.

Fig. 15 

PCM Switchover 

This is a similar to T.38 Fax Relay technology of the data transfer via IP-gateway with the usage of usual analog modems. When detecting a signal with a carrier frequency of an analog modem at the moment of the phase of data transfer there is created a transparent IP-connection with usage of G.711 codec (64 Kbit/s). With V.90 modem, good connection with a local operator of IP-telephony and PCM Switchover support you can get the connection similar to 64 Kbit/s channel. 

Conclusion

  1. Date rate and a bandwidth of an IP-channel in the network under H.323 can't be defined on the assumption of only a coding speed of the utilized codecs and the number of simultaneous conversations. They also depend on the number of enquiry messages of abonents for the connection at the current moment, IP-packet structure, the voice detector threshold, methods of user authorization, the number of working in the network gatekeepers, and many others. 
  2. Summarizing the results, we should say that a clear channel with 256 Kbit/s bandwidth and a gateway with one connection to the telephony network through the E1 digital path is the most interesting alternative for network organization for a beginning ITSP. 
  3. Depending on what the potential abonent needs, the operator must define the network structure, find a partner for a traffic termination, define the equipment, gateway connection type etc. 

Besides: 

  1. If you are building a network with usage of n? 64 Kbit/s clear channels, we can recommend you to use G.729a (8 Kbit/s) voice codec or G.723.1 (6,3 Kbit/s), or even sometimes G.711. If you are going to send faxes, we'd advise to include T.38 Fax Relay with the speeds up to 14,4 Kbit/s. For those, who want to utilize voice modems, we'd recommend to include support for PCM Switchover and make a separate tariff plan. 
  2. In IP-gateways which are based on the infrastructure of the Public Internet, it's better to use G.723.1, G.729b, NetCoder codecs. The speed of fax transfer with the usage of T.38 Fax Relay should be decreased to 9,6 Kbit/s. 
  3. VAD technology will help to economize the bandwidth in a voice gateway, though with some voice quality degradation. Unfortunately, not all voice gateways allow to adjust the voice detector threshold and some devices has a fixed value equal to 30%. 
  4. Not all gateways with support for Fax Relay T.38 are compatible. It's because of ambiguity of methods of document image transfer. That's why, if you are a beginner, we'd first recommend you to find out the type and characteristics of the equipment of the supposed Partner. 

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