Patent Publication Number: US-2009219939-A1

Title: Transporting Packets

Description:
This application claims priority on British Patent Application No. 0602314.7, filed Feb. 6, 2006, the disclosure of which is fully incorporated herein by reference. 
     FIELD OF THE INVENTION 
     The present invention relates to a method of transporting packets, in particular to transporting packets between one network node and another network node. 
     BACKGROUND OF THE INVENTION 
       FIG. 1  is a schematic illustration of devices  11 , 12 , 13  in a first network, Network  1 , communicating with devices  21 , 22 , 23  in a second network, Network  2 . For example, device  11  in Network  1  may be sending packets to devices  21  and  23  in Network  2 , and device  12  in Network  1  may be sending packets to device  22  in Network  2 . Packets sent by a device  11 , 12 , 13  in Network  1  are initially routed to a host  1  in Network  1 , and are then sent over a communication link, for example, an IP backbone  3 , to a host  2  in Network  2 . The host  2  in Network  2  routes a received packets to device  21 , device  22  or device  23  in accordance with the destination specified in the packet. All packets passing from Network  1  to Network  2 , or vice versa, pass over the link  3 . 
     A packet consists, in general, of a header and a payload. The header contains details of the routing of the packet, such as the destination of the packet and the origin of the packet. Thus, for example, a packet generated by device  11  and destined for device  21  will have a header that, inter alia, identifies the destination of the packet to be device  21 . When the packet is received at the host  1 , the host  1  will read the destination from the header of the packet, will identify the destination as being in Network  2  and will forward the packet to the host  2  of Network  2  over the link  3 . When the packet is received at the host  2  of Network  2 , the host  2  will read the destination from the header of the packet and will identify the destination as device  21 . 
     The address for a device (or, more generally, for a “network node”) is commonly an IP (Internet Protocol) address. 
     A particular device may serve a number of different applications such as, for example, a e-mail system, a web browser, etc. The header of a packet therefore generally not only identifies the destination device but also identifies the particular application to which the packet&#39;s contents relate. The particular application is generally identified by a “UDP port number”. The header of a packet transmitted along link  3  will therefore typically contain an IP address field and a UDP port field. 
     The header of a packet may, depending on the application to which the packet&#39;s contents relate, contain further fields such as, for example, a “Real Time Protocol” (RTP) field. An RTP field relates to the framing and handling of real time data such as VoIP data. Thus, header of a VoIP packet transmitted along link  3  will therefore typically contain an RTP field in addition to an IP address field and a UDP address field. 
     There is currently much effort expended on transmitting telephone calls over the Internet, or “VoIP” (Voice over Internet Protocol). One major issue in VoIP is the massive amount of overhead caused by IP/UDP/RTP framing of the packet header. With IPv6 (Internet Protocol version 6 address realm) a header with IP, UDP and RTP fields has a total length of 60 bytes but the actual payload of the packet may be as low as 15-20 bytes, meaning that over 75% of the bandwidth may be consumed by the header of the packet. 
     Several header compression schemes have been introduced to overcome this problem, but they have been designed for point-to-point connections rather than for a case where many different packet flows are transmitted over a common link as in  FIG. 1 . These header compression schemes are in practice useless except in the radio interface. Moreover, if header compression were introduced in a fixed network it would have strict requirements for compression which it would not be possible to follow in a public commercial IP network. Thus there is a need for other means to decrease the bandwidth demand of VoIP in a core network according to the 3GPP specifications. 
     SUMMARY OF THE INVENTION 
     A first aspect of the present invention provides a method of transporting IP packets between nodes of an IP network, the method comprising:
         receiving packets at a first network node;   separating the received packets into a plurality of streams, where the packets of each stream share a common IP header;   merging a plurality of packets to form a merged packet; and   sending the merged packet to a second network node.       

     In the method of the invention two, or more, packets are merged (or are “multiplexed”). The payloads of the component packets used to form the merged packet are sent in the merged packet under a common header. The header of the merged packet occupies a lower proportion of the length of the merged packet than does the header of a single packet. That is if two packets (as an example) are merged, the length of the header of the merged packet will be less than the sum of the lengths of the headers of the two component packets. 
     In the case of real-time traffic, a merged packet preferably contains no more than one packet from any stream. In the case of non real-time traffic, however, it is in principle possible for a stream to contribute two or more packets to a merged packet. 
     Sending a merging packet through a communications network requires no modification to the network. 
     When the merged packet is received at the second network node, it is separated into its component packets. 
     A method of the invention may be applied to, for example, speech traffic transported over IP in a 3GPP UMTS network over an Nb-interface between media gateways (MGWs) or over an Iu-interface between an RNC and MGW. However, merging (or multiplexing) packets can be performed essentially for all packets heading to the same IP address, and the invention can be used for all kind of UDP traffic. A multiplexing method of the invention may in particular be used with RTP packets (although it may not be applicable to RTCP (Real Time Control Protocol) which may continue to be transported normally by IP/UDP packets. The multiplexing method is independent of the protocols beneath IP and it can be used in, for example, an MPLS (Multi-Protocol Label Switching) enabled network as well as in any other IP-based network. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Preferred embodiments of the invention will now be described by way of illustrative example with reference to the accompanying figures in which: 
         FIG. 1  is a schematic illustration of a communications system; 
         FIG. 2  is a block flow diagram of a method of the invention; 
         FIGS. 3(   a ) and  3 ( b ) are schematic illustrations of packets used in the method of the invention; 
         FIG. 3(   c ) is a schematic view of a multiplex header; 
         FIG. 3(   d ) is a schematic illustration of another packet used in the method of the invention; 
         FIG. 4  is schematic illustration of formation of a merged packet; 
         FIG. 5  is a schematic view of a multiplex header; and 
         FIG. 6  is a schematic view of a compressed RTP header. 
     
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     The invention will be described, by way of example, with reference to the network shown in  FIG. 1 . It will be assumed, as an example, that various applications controlled by devices in Network  1  are sending packets to devices Network  2 , but the invention may equally well be applied to the case of devices in Network  2  sending packets to Network  1 . 
     In one embodiment, one device in Network  1 , for example device  11 , is the source of two or more streams of packets destined for devices in Network  2 . It is assumed that the streams have the same DiffServ class. This in practice requires that the traffic of one stream is very similar to that of the other stream, and in many cases the two streams are likely to relate to the same application. 
     It is further assumed that the addresses are already negotiated between the devices so that, for example, a packet going from device  11  to  21  has device  11 &#39;s IP address as the source address and device  21 &#39;s IP address as the destination address. In this example, packets in a stream going from device  11  to device  21  can be merged only with packets in one or more other stream going from device  11  to device  21  and multiplexing for those streams is performed in device  11 . (Similarly, merging of streams going from device  12  to device  21  would be performed in device  12 , merging of streams going from device  21  to device  11  would be performed in device  21 , and so on.) Hosts  1  and  2  are used only as routers in this example. 
     In this example, initially packets are received at device  11  (step  1  of  FIG. 2 ). In this example, these are assumed to include, inter alia, two or more streams of packets destined to device  21  and that the same DiffServ class (and may also include further streams of packets having a different destination device in Network  2  and/or relating to another DiffServ class). Packets destined for device  21  will have a header containing, inter alia, the IP address of device  21 , and packets destined to device  22 [ 23 ] will have a header containing, inter alia, the IP address of device  22 [ 23 ]. 
     Next, the packets received at device  11  are sorted into two or more streams, with each stream corresponding to a single IP address (step  2  of  FIG. 2 ). In this example, this sorting step will result in two (or more) streams of packets destined from device  11  to device  21  and that relate to the same DiffServ class, and may possibly also result in further streams of packets having a different destination device in Network  2  and/or relating to another DiffServ class. 
     It is preferable that each stream created at step  2  contains only packets of one type (more formally, each stream preferably contains only packets of one DiffServ class). Thus, in the example in which two (or more) different applications (for example VoIP and e-mail) controlled by device  11  in Network  1  are sending packets to device  21  in Network  2 , for example, the packets arriving at device  11  would preferably be sorted into streams such that VoIP packets and e-mail packets were not included in the same stream as one another. 
     Next, a plurality of packets are selected (step  3  of  FIG. 2 ) and are merged to form a merged packet (step  4  of  FIG. 2 ). The packets selected for merging have the same IP address as one another, for example are all destined to device  21  and so have device  21 &#39;s IP address as their destination address. The merged packet contains a single IP field in its header, since all the component packets of the merged packet are destined for the same IP address. The merged packet also contains a plurality of payload fields, one payload field for each of the component packets. 
     In the case of real-time traffic, a merged packet preferably contains no more than one packet from any stream. In this case, the selecting step would consist of selecting one packet from each stream (or, more generally, selecting one packet from each stream that is not empty). If a stream contains two or more packets, the packet that is selected is preferably the packet that has been waiting in the stream for the longest time. 
     In the case of non real-time traffic, however, it is in principle possible for a merged packed to contain two or more packets from a stream. 
     In a preferred embodiment, a merged packet contains only packets of the same type (that is, contains packets of only one DiffServ class). In this embodiment a merged packet may be formed by merging a packet from one stream of, for example, VoIP packets destined for device  21  with a packet from another stream of VoIP packets destined for device  21 . Similarly, a merged packet may be formed by merging a packet from one stream of, for example, e-mail packets destined for device  21  with a packet from another stream of e-mail packets destined for device  21 , by merging a packet from one stream of, for example, VoIP packets destined for device  22  with a packet from another stream of VoIP packets destined for device  22 , etc. 
     Next, the merged packet is transmitted to the destination indicated by the IP address field in the header of the merged packet (step  5  of  FIG. 2 ). In the example, a merged packet formed by merging a packet from two streams of VoIP packets destined for device  21  is transmitted to device  21 , a merged packet formed by merging a packet from two streams of packets destined for device  22  is transmitted to device  22 , and so on. 
     When the merged packet is received at the destination indicated by the IP address field in the header of the merged packet, the merged packet is split into its component packets. 
     In a preferred embodiment of the invention, the header of the merged packet contains an indication that the packet is a merged packet. In a particularly preferred embodiment, this is done by including, in the header of the merged packet, a UDP port number that denotes that the packet is a merged packet. This UDP port number corresponds to a UDP port at the destination device that is reserved for merged packets—one UDP port is assigned as the “multiplexing port” into which all merged packets are sent. Since this port is reserved for merged packets, it is not allowed to be assigned to any individual connections and this port will thus receive only merged packets. The destination device then knows when an arriving packet is a merged packet. In this embodiment, when a merged packet is received at the destination device indicated by the IP address field in the header of the merged packet, the merged packet would be directed to the UDP port of the destination device that is reserved for merged packets. 
     By including a UDP port number in the header of the merged packet, no additional mapping tables are required. 
     The UDP port reserved for merged packets is proposed to be port  2002 . 
       FIG. 3(   a ) is a schematic illustration of a merged packet  14  used in the method of the invention. The packet  14  has, as usual, a header  4  and a payload  5 . The header contains an IP address field  6 , which contains the IP address of the destination device of the merged packet. As explained above, the header  4  may also contain a UDP field  7  that identifies the port number of the UDP port reserved for merged packets at the destination device. Alternatively, the header may contain another field that identifies the packet as a merged packet. 
     For non-real-time traffic it is possible for all component packets used to form the merged packet to have the same UDP port number, and in this case the UDP field  7  of the header  4  could contain the original UDP port number rather than a UDP port number that is specifically assigned to merged packets. This would, however, require that all UDP ports would have to be able to detect and handle merged packets. In practice, therefore, it is preferable have a UDP port, which is distinct from the conventional application UDP ports, specifically assigned to handle merged packets. If one UDP port is assigned to handle merged packets, all other ports can receive packets normally and are not affected by the packet multiplexing of the invention. 
     The payload  5  of the merged packet  14  contains an number of multiplexed packet fields  5   a , 5   b , one multiplexed packet field corresponding to each component packet.  FIG. 3(   a ) shows a merged packet formed by merging two component packets, and the merged packet therefore has two multiplexed packet fields; however, the invention is not limited to this and more than two packets may be merged to form a merged packet. 
     The header  4  of the packet  14  is a “common header”, in that the information in the header  4  is common to all the multiplexed packet fields  5   a , 5   b.    
     Each multiplexed packet field  5   a , 5   b  contains a multiplexing header (“MUX header”)  8   a , 8   b  and a payload field  9   a , 9   b . The multiplexing header is shown in  FIG. 3(   c ). The multiplexing header  8   a , 8   b  contains a multiplexing ID field  15 , which may include the whole UDP port information. The multiplexing ID field  15  is for identification of different connections. Its value is the same as the UDP destination port if the packet were sent conventionally without using the multiplexing of the invention—that is, the multiplexing ID field  15  is the UDP port number of the original component packet. 
     The multiplexing header  8   a , 8   b  preferably also contain a payload length indicator field  16 . This field is provided since the multiplexing ID field  15  does not indicates when the next multiplexed packet field starts. Thus, the payload length indicator field  16  indicates the number of bytes in the each multiplexed packet field (header  8   a , 8   b  and payload  9   a , 9   b ) to enable the start of the next multiplexed packet field to be determined. 
     In one embodiment, the multiplexing ID field is a 16-bit (2-byte) field, and the payload length indicator field is an 8-bit (one-byte) field (giving a maximum length for the fields following the payload length indicator field of 256 bytes). Thus, in this embodiment, the MUX header field, overall, is a 3-byte field. 
     Each multiplexed packet field  5   a , 5   b  is formed as: 
       ∥Mux ID/UDP port (2 bytes)|Length Indicator (1 byte)|Payload (1-256 bytes) 
     The header  4  of the merged packet  14  may contain other fields in addition to those shown in  FIG. 3(   a ). As an example, the header  4  of the merged packet may further contain an RTP header field that contains RTP information for the merged packet.  FIG. 3(   b ) is a schematic illustration of a merged packet  14 ′ whose header  4  further contains an RTP header field  10 . 
     In some applications, such as voice traffic in a 3GPP network, the RTP information is essential. The packet structure shown in  FIG. 3(   b ) may be used where all the component packets of the merged packet share a common IP/UDP/RTP header and could be used in cases where individual RTP information for each packet is not needed. In some applications, however, individual RTP information for each packet is needed and, in such a case, the packet structure of  FIG. 3(   b ) is not suitable. 
       FIG. 3(   d ) shows a further packet structure in which the header  8   a , 8   b  of each multiplexed packet field  5   a , 5   b  contains RTP information. The RTP information is provided in an RTP field  17   a , 17   b  within the MUX header  8   a , 8   b . In this example, the RTP field  17   a , 17   b  has a length of 12 bytes. 
       FIG. 4  illustrates formation of the merged packet of  FIG. 3(   d ). The common header  4  of the merged packet contains an IP address field  6  and a UDP header field  7 . The UDP header field contains a UDP port number (eg 2002) identifying the packet as a merged packet. Each multiplexed packet field  5   a , 5   b  has an MUX header containing a UDP port number corresponding to the UDP port number of the component packet, and a payload length indicator field (not shown). Each multiplexed packet field  5   a , 5   b  further has an RTP header field  17   a , 17   b , and a payload field  9   a , 9   b  (in this example, the payload is shown as an NbUP frame). The fields following the payload length indicator field—that is, the RTP header field  17   a , 17   b , and a payload field  9   a , 9   b —have a maximum overall length of 256 bytes (in the example of a one byte payload length indicator field). 
     It is preferable that packets are selected for merging at step  3  of  FIG. 2  soon after they have been put into their respective streams, to avoid introducing an additional delay into the network. In one embodiment, the merging step is carried out by selecting packets that arrived at the network node within a given time frame (and forming the merged packet soon after the end of the time frame). The duration of the time frames should be made large enough to ensure that packets are likely to have arrived in several streams in a time frame (carrying out the merging step by selecting packets that arrived at the network node within a given time frame effectively limits the number of component packets merged into a merged packet), to ensure that a useful saving in bandwidth is obtained, but should not be so large that significant delay and multiplexing jitter are introduced. A duration of around 1 ms for the time frame should be suitable in many applications—a time frame of this order should be long enough to gather several packets into a merged packet but should be short enough to keep delay and multiplexing-jitter low. 
     For applications that are not time-sensitive, or if a user is prepared to accept greater delay/jitter to obtain greater bandwidth savings, a longer time frame can be used. A duration of 1 ms is, however, a preferred value for real-time traffic. 
     Subject to the desire not to introduce significant delay into the network, there is essentially no limitation on the number of packets that can be multiplexed into a merged packet. An IP datagram has a maximum length of 65535 bytes and Ethernet frame has a maximum length of 1518 bytes, meaning that the Ethernet frame size is, in practice, what limits the number of packets that can be multiplexed into one merged packet. 
     A further benefit of the present invention, in addition to the reduction in bandwidth, is that the number of packets in the network is also reduced. When only two packets at a time are multiplexed into a merged packet, the number of packets in the network is immediately reduced to 50% of its previous value. Moreover, tens of packets may be multiplexed into one merged packet, and this means that the number of packets in the network may be reduced to a few percent of the number of packets in a network when multiplexing is not used. (If ten packets at a time, for example, are multiplexed into a merged packet, the number of packets in the network is reduced to 10% of its previous value.) This reduction in number of packets in the network brings tremendous processing savings in the network routers, which process traffic per packet. This has a positive influence also to speech quality, in the case of VoIP, since fewer packets are in the network and therefore fewer packets are likely to be dropped during transmission. If a packet is dropped for some reason the impact spreads out to multiple connections and the actual impact on one connection is not, in practice, substantial. 
     Table 1 shows examples of the bandwidth reduction that can be obtained by the present invention. Table 1 shows results for four different cases, PoS for both IPv4 and IPv6, and Ethernet for both IPv4 and IPv6. In the PoS examples the network is assumed to use double MPLS framing (VPN and traffic type differentiation) and the Ethernet examples assume that a VLAN tag is used. The figures are for AMR12.2 packets with a 60% activity factor. 
     Table 1 shows, for each of the four cases, the bandwidth (BW) without multiplexing, the bandwidth with 2 packets multiplexed into a merged packet, and the bandwidth with 10 packets multiplexed into a merged packet. The results assume that the merged packets have the form shown in  FIG. 3(   d ), with a common IP/UDP header and RTP information in each MUX header. 
     
       
         
           
               
               
               
               
               
             
               
                   
                 TABLE 1 
               
               
                   
                   
               
               
                   
                 PoS, IPv4 
                 PoS, IPv6 
                 Eth, IPv4 
                 Eth, IPv6 
               
               
                   
                   
               
             
            
               
                   
               
            
           
           
               
               
               
               
               
            
               
                 BW ref 
                 22.88 kbps 
                 28.08 
                 29.90 
                 35.10 
               
               
                 BW, 2 pkts 
                 18.07 kbps 
                 20.67 
                 21.58 
                 24.18 
               
               
                 Decrease 
                 21% 
                 26% 
                 28% 
                 31% 
               
               
                 BW, 10 pkts 
                 13.60 kbps 
                 14.12 
                 14.30 
                 14.82 
               
               
                 Decrease 
                 41% 
                 50% 
                 52% 
                 58% 
               
               
                   
               
               
                 (The bandwidths shown for 2 or 10 component packets merged into one are the effective bandwidth per component packet - i.e. a merged packet having two component packets would, in the PoS, IPv4 case, have a bandwidth of 36.14 kbps.) 
               
            
           
         
       
     
     In a further preferred embodiment of the invention, to achieve even better bandwidth savings, the RTP header fields are compressed. This is possible since an RTP header includes many static fields that remain unchanged during an RTP session, so that in many cases the RTP header may be compressed. The RTP compression is additional to the multiplexing of packets into merged packets, so that it is always possible to go back to pure multiplexing in circumstances where RTP compression may not be used. 
     In a particularly preferred embodiment, a merged packets that use RTP header compression are sent to a UDP port that is reserved for merged packets with RTP header compression. Thus, preferably one UDP port that is reserved for merged packets with RTP header compression and another UDP port is reserved for merged packets without RTP header compression. 
     The UDP port reserved for merged packets with RTP header compression is proposed to be port  2004 . This port would used in the same way as port  2002  is used for merged packets without RTP header compression. 
     In compression there is always an initialization phase first where the full header is transferred to the receiver. The full header is stored and it is used in decompression. After the initialization phase, only compressed headers are sent unless the information in the header changes too much in which case it would be necessary to send a full header. Alternatively, to confirm that the full header has been received and the initialization is done the sending of the header compressed packets may not be initiated immediately after the first packet. For example, the first ten packets may be non-compressed packets with subsequent packets being compressed packets, and the same procedure (ie, sending the first ten packets as non-compressed packets) may be repeated if the header has to be updated. 
     One possible structure for the header  8   a , 8   b  of a multiplexed packet field is shown in  FIG. 5 . The multiplexing header of  FIG. 4  includes a type field  18 , having a length of 1 bit. The type field  18  has two possible states, 0 for indicating a merged packet without RTP header compression and 1 for indicating a merged packet with RTP header compression. 
     The header  8   a  of  FIG. 5  also includes a multiplexing ID field  15  and a payload length indicator field  16 . The multiplexing ID field  15  has a length of 15 bits, and is for identification of different connections. The value of the multiplexing ID field  15  is the same as the UDP destination port of a non-multiplexed packet divided by two (only even numbered ports are used for RTP sessions). The payload length indicator field  16  has a length of 8 bits, and gives the length (header+payload) of the multiplexed RTP packet in bytes. 
       FIG. 6  illustrates one possible format of a compressed RTP header  17 ′. The RTP header compression mechanism of  FIG. 6  is an example and other mechanisms may be also used. 
     An RTP header contains two fields that change during a connection and that need to be transferred within each packet, the sequence number and timestamp. Both fields change, however, in a well defined way. The sequence number steps by one with every sent packet indicating any packet losses, and the timestamp depicts the time difference between consecutive packets. 
     The compressed RTP header  17 ′ of  FIG. 6  includes a sequence number (SN) field  19 , of length 3 bits. The sequence number field  19  changes as the original sequence number, but has only 8 states, which should be enough since packets are generally sent in very low BER networks. If the sequence number field is inadequate, a full RTP header should be used. 
     The compressed RTP header  17 ′ of  FIG. 6  also includes a timestamp (TS) field  20 , of length 5 bits. The timestamp field  20  changes basically as the original timestamp, but the actual time difference denoted by one step of the timestamp field depends on the payload type since the type is known based on the initialization messages. One step of the timestamp field  20  in the compressed RTP header  17 ′ signifies  80  steps (5 ms×16 kHz=80) for PCM voice and 320 steps (20 ms×16 kHz=320) for AMR coded voice when converted to the original steps in a timestamp field. If the timestamp field is for some reason inadequate, a full RTP header should be used. 
     The compressed RTP header  17 ′ may be used in place of the full (ie, uncompressed) RTP header fields in a merged packet, for example in place of the full RTP header fields  17   a , 17   b  of the merged packet of  FIG. 3(   d ). 
     Alternatively, the in the compressed RTP header of  FIG. 6  the sequence number SN field may be 8 bits, and the timestamp TS field may be 16 bits. In this case each field has half the length that it would have in an uncompressed header according to Request for Comments 3550. The TS field changes as the original timestamp (RFC3550) but the length is half of the original resulting in modulo of 4 seconds with 16 kHz clock reference. 
     Table 2 shows examples of the bandwidth reduction that can be obtained by the present invention when RTP compression is employed. Table 2 shows results for the four cases considered in Table 1, under the same assumptions as made for Table 1. 
     
       
         
           
               
               
               
               
               
             
               
                   
                 TABLE 2 
               
               
                   
                   
               
               
                   
                 PoS, IPv4 
                 PoS, IPv6 
                 Eth, IPv4 
                 Eth, IPv6 
               
               
                   
                   
               
             
            
               
                   
               
            
           
           
               
               
               
               
               
            
               
                 BW ref 
                 22.88 kbps 
                 28.08 
                 29.90 
                 35.10 
               
               
                 BW, 2 pkts 
                 15.73 kbps 
                 18.33 
                 19.24 
                 21.84 
               
               
                 Decrease 
                 31% 
                 35% 
                 36% 
                 38% 
               
               
                 BW, 10 pkts 
                 11.26 kbps 
                 11.78 
                 11.96 
                 12.48 
               
               
                 Decrease 
                 51% 
                 58% 
                 60% 
                 64% 
               
               
                   
               
            
           
         
       
     
     It should be noted that not all devices can support multiplexing to form merged packets and/or also RTP header compression. Thus, before sending packets it is necessary for the sending node and the receiving node to agree that multiplexing to form merged packets is used, and also to agree whether RTP header compression is used. 
     The bearer initialization phase in both Nb- and Iu-interfaces include mandatory messages for the support mode that is used, e.g. for speech traffic. Nb/Iu UP PDU Type 14 is used at initialization and the messages include spare extension fields (in both initialization and acknowledgement frames) that can be used for any additional function. It is proposed that these fields are used to detect whether a multiplexing method of the invention is applicable. (The transparent mode in the Iu-interface would not support multiplexing since it has no initialization phase, but this mode is not used by real-time applications.) 
     When an originating node (e.g., a MGW (media gateway) or RNC) that supports multiplexing sends an initialisation message it would indicate in the initialisation message that it would like to use multiplexing, e.g. by setting the first bit to 1 or using a specific sequence in the spare extension field of the initialization frame. The receiving node knows, from that bit or sequence in the initialisation message, that multiplexing can be used. If the receiving node supports multiplexing it copies the bit or sequence to the spare extension field of the positive acknowledgement message for the indication, ands transmits the acknowledgement message to the originating node. This confirms to the originating node that multiplexing can be used. If, however, the receiving node does not support multiplexing it just ignores the spare extension in the initialization message and sends a regular acknowledgement—the originating node that started the initialization knows then not to use multiplexing. 
     Since a node may support multiplexing but not RTP header compression, the initialisation message sent by an originating node that can support both multiplexing and RTP header compression must contain separate indications relating to multiplexing and RTP header compression. For example, if the first bit of the initialisation message indicates that the originating node can support multiplexing, then the fact that the originating node can additionally support RTP header compression could be indicated by the first two bits (or by a different sequence). The destination node can now reply in three ways. The destination node may indicate that multiplexing with RTP header compression may be used, for example by repeating the bits or sequence. The destination node may however reply by indicating that multiplexing but not RTP header compression may be used (for example by replying with the one bit/sequence indicating normal multiplexing) or it may reply by indicating that multiplexing may not be used (for example, by replying without any multiplexing indications). 
     For an Nb interface there already is a standardised protocol for bearer control, IP Bearer Control Protocol (IPBCP), and this could be used also for detecting multiplexing applicability. IPBCP cannot however be used for an Iu-interface and therefore UP initialization as a more common solution may better for initializing when it is necessary to determine whether multiplexing is possible. In general, the step of determining whether multiplexing is applicable can be seen as a migration phase function, which could be left out when it is known that all relevant nodes support multiplexing. In this case, multiplexed packets could be always detected based on the UDP port (e.g., port  2002  for normal multiplexed packets and port  2004  for multiplexed packets with RTP header compression). 
     In principle, RTP header compression could be applied to a merged packet having a common RTP header (as in  FIG. 3(   b )), or even to a conventional unmerged packet. In these cases, however the bandwidth saving with RTP header compression would be only about 5-10%, which is unlikely to justify the complexity that RTP header compression would add to the connection handling. 
     In the example given above, it was assume that addresses had been negotiated between devices so that, for example, a packet going from device  11  to  21  has device  11 &#39;s IP address as its source address and device  21 &#39;s IP address as the destination address. In this example, only streams from one originating device  11 , 12 , 13  to one terminating device  21 , 22 , 23  may be merged, and the merging occurs in the originating device. Hosts  1  and  2  are used only as routers in this example. The invention is not however limited to this. 
     For example, it may be that networks  1  and  2  have some other internal routing mechanism and that for all streams between a device  1   x  and a device  2   x  the connection between host  1  and  2  has to be separately established. (“Device  1   x ” denotes the devices  11 ,  12 ,  13  in Network  1 , and “device  2   x ” denotes the devices  21 ,  22 ,  23  in Network  2 .) This is the case in e.g. 3GPP based networks where an IP transport network is used between host  1  and  2 , and networks  1  and  2  are in practice radio access networks which use TDM or ATM instead of IP. The hosts  1  and  2  then are media gateways that handle media conversions and negotiations so that network  1  can be connected to network  2  (since direct communication as in the previous example is not possible owing to the different transport network in between). Since the connection between host  1  and  2  is negotiated separately and is independent of the real source and destination, all streams of packets going from network  1  to network  2  have, at link  3 , the same IP source &amp; destination address pair and they can all be merged (multiplexed) in the Host  1  for packets going from Network  1  to Network  2  (and can be multiplexed in the Host  2  in the case of packets going from Network  2  to Network  1 ). 
     In this alternative example, packets arriving at host  1  (in the case of packets passing from Network  1  to Network  2 ) are sorted into streams, with each stream preferably comprising packets of a single DiffServ class. A merged packet is then formed by merging packets from two or more streams, and the merged packet is transmitted to Host  2 . The merged packet is de-merged in Host  2 , and the constituent packets are passed to their respective destinations in Network  2 . 
     This second example particularly shows the potential of the mechanism since the multiplexing gain is directly proportional to the number of streams that can be multiplexed and now the merging can be performed at the link where the traffic load is normally the highest and the most expensive (ie, the core links between separate sites). When the merging is performed at Host  1 , it is possible to merge a packet from a stream originating at device  11  and destined for device  21  with a packet from any other stream originating at a device  1   x  in Network  1  and destined for a device  2   x  in Network  2  (subject to the proviso that preferably packets of only one DiffServ class are merged). 
     The method of the invention may be applied to IMS packets. This would require a different kind of initialization process, with SIP. 
     The method of the present invention may be implemented by a suitably programmed processor. A program for controlling a processor to perform the method of the invention may be stated on any suitable storage medium such as, for example, a computer disk, a magnetic disk or an optical disk.