packet

问题 When will a TCP packet be fragmented at the application layer? When a TCP packet is sent from an application, will the recipient at the application layer ever receive the packet in two or more packets? If so, what conditions cause the packet to be divided. It seems like a packet won't be fragmented until it reaches the Ethernet (at the network layer) limit of 1500 bytes. But, that fragmentation will be transparent to the recipient at the application layer since the network layer will

问题 I asked earlier about H264 at RTP H.264 Packet Depacketizer My question now is about the audio packets. I noticed via the RTP packets that audio frames like AAC, G.711, G.726 and others all have the Marker Bit set. I think frames are independent. am I right? My question is: Audio is small, but I know that I can have more than one frame per RTP ​​packet. Independent of how many frames I have, they are complete? Or it may be fragmented between RTP packets. 回答1: The difference between audio and

问题 How can I pull relevant packet information from a JpCap packet? I am trying to find the source/destination ip and port. So far I have tried string parsing the Packet's toString() method, but that seems brutish. 回答1: You have to cast the Packet object to the correct type of Packet i think. So something like: TCPPacket p = (TCPPacket)packet; // Get the tcp src and dest ports int destPort = p.dst_port; int srcPort = p.src_port; // Get the src and dest IP addresses from the IP layer InetAddress

问题 I am new to web services so I created a web service to replace my current in-app DB transactions. I wanted things to be safer so that is why I went this way. When using a free packet sniffer, it instantly grabs my web service ASMX call. The problem with this is that using a program such as fiddler they can easily see the data going back and forth and even worse set up a auto responder. Is there a way to hide the calls being sent to the web service to hide from packet sniffers? Or at least

问题 Is it possible to have libpcap remove a packet instead of just sniff it as it passes through? I'm wanting to intercept each packet and encapsulate it into a new packet along with measurement data, but both packets (mine and the original) both reach the destination. 回答1: It's not possible. You need to write a driver (for your operating system) to make the networking stack filter out packets. 回答2: The only way you could do this is by being the only physical path between the sender and receiver

问题 I'm using libpcap in C++ for reading packets from pcap files, e.g.: rc = pcap_next_ex((pcap_t*)handle, &header, (const unsigned char**)packet); I would like to parse the packets header (without the payload). For example, how can I parse a given packet in order to extract its source and destintation ip addresses? thanks 回答1: Checkout the code sample for libpcap http://www.tcpdump.org/pcap.html In the got_packet(u_char *args, const struct pcap_pkthdr *header, const u_char *packet); function you

问题 I've been learning error correction mechanisms for very noisy AWGN channels. I've read about Hybrid ARQ systems that use FEC coding in conjunction with ARQ systems which sounds like a good starting point for these type of channels. I was thinking of a setup that follows the next example: First transmission of Packet #1 - RS(38,8) Received a NAK Second transmission of Packet #1 - additional FEC bytes such that at the receiver it gets RS(46,16) Received ACK My question is, is it possible to pre

问题 I wrote this piece of code to get http header and set Host: http_layer = packet.getlayer(http.HTTPRequest).fields http_layer['Host'] = "newHostName" return packet After running the afforementioned code,the new host name has been set correctly, but the problem is that when I write the packet in pcap file, I still see the previous host in http fields, Is there an absolute way to manipulate http_layer['Host'] ? Any help would be appreciated. Regards. 回答1: After all, found the answer. The key is

问题 I need to know exactly how many packets per interval my BLE can handle. The peripheral, as per its datasheet, handles 6 packets per interval, but I have been unable to find out how many packets the Central can handle. The Central device is a Motorola Moto G (generation 2), running Android 5.0.2. By examining he btsnoop_hci.log file I have been able to identify multiple connection parameters, such as the connection interval (7.5ms in my case). My questions is wheter it is possible to determine

问题 I'm trying to implement a simple data transfer using UDP. I have a problem for the checksum, given a packet containing the data, how should I implement the checksum? also any idea how to implement the timeouts so it will trigger the retransmission ? Thanks 回答1: Why not try Reliable UDP, see http://en.wikipedia.org/wiki/Reliable_User_Datagram_Protocol It has a standard. 回答2: here's one approach for the internet checksum unsigned short checkSum() { unsigned long sum = 0; int i; for(i=0; i <