在点点滴滴中进步；在细致细微处成长！
Idle Speech Frame/No_Data
-------------------------------------------
no speech(FR) Bad Frame(FR)
Idle Speech Frame(FR)
------------------> -----------------> -------------->
MS BTS BSC MSC
<------------------ <----------------- <--------------
Idle Speech Frame(FR) no speech(FR)
&
HSCSD：High-Speed Circuit-Switched Data
- HSCSD使用不同的编码方式和多时隙传输来提高速度
- 不同的误码校正方式
- CSD为了适应最烂的网络环境，需要使用冗余很大的编码方式以便作误码校正
- HSCSD提供不同的误码校正级别
- 最好情况下，HSCSD单时隙从CSD的9.6 kbps提升到14.4 kbps
- 最坏情况下，等于CSD的速度
- 多时隙传输
- 4个时隙，最好情况下 4 * 14.4 = 57.6 kbps，最坏情况下 4 * 9.6 = 38.4 kbps
- 8个时隙，最好情况下 8 * 14.4 = 115 kbps
- HSCSD需要这些时隙完全被一个用户占用
- 这在通话开始，或通话进行中都是可能的
- 并非用户的全部请求都可以被满足，因为语音通话优先级更高
- HSCSD比CSD贵，没有GPRS竞争力强
- 占用更多的时隙，当然多交点钱了
- 这也是GPRS比HSCSD流行的原因
- HSCSD相比GPRS的优势之一就是无线接口平均等待时间短，因为不需要网络等待权限判断再发数据包
- HSCSD也是EDGE和UMTS的选项
- 相比接入快速且高带宽的UMTS系统来说，HSCSD主要用在对付只能拨号的系统
6.3.4 Analyzing sender and receiver reports
It is expected that reception quality feedback will be useful not only for the sender but also for other receivers and third-party monitors.The sender may modify its transmissions based on the feedback; receivers can determine whether problems are local, regional or global;network managers may use profile-independent monitors that receive only the RTCP packets and not the corresponding RTP data packets to evaluate the performance of their networks for multicast distribution.
Cumulative counts are used in both the sender information and receiver report blocks so that differences may be calculated between any two reports to make measurements over both short and long time periods, and to provide resilience against the loss of a report. The difference between the last two reports received can be used to estimate the recent quality of the distribution. The NTP timestamp is included so that rates may be calculated from these differences over the interval between two reports. Since that timestamp is independent of the clock rate for the data encoding, it is possible to implement encoding- and profile-independent quality monitors.
An example calculation is the packet loss rate over the interval between two reception reports. The difference in the cumulative number of packets lost gives the number lost during that interva
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P| RC | PT=SR=200 | length |header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| NTP timestamp, most significant word |sender
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+info
| NTP timesta
Definetion - In VoIP, a jitter buffer is shared data area where voice
packages can be collected, stored, and sent to the voice processor in
evenly spaced intervals. Variations in packet arrival time, called jitter,
can occur because of network congestion, timing drift, or route changes.
The jitter buffer, which is located at the receiving end of the voice
connection, intentionally(故意) delays the arriving packets so that the end
user experiences a clear connection with very little sound distortion(失真).
There are two kinds of jitter buffers, static and dynamic. A static
jitter buffer is hardware-based and is configured by the manufactures.
A dynamic jitter buffer is software-based and can be configured by the
network administrator to adapt to changes in the network's delay.
Problem: Jitter Buffer
A jitter buffer temporarily stores arriving packets in order to minimize
delay variations(变化). If packets arrive too late then they are discarded.
A jitter buffer may be mis-configured and be either too large or too small.
Impact
If a jitter buffer is too small then an excessive number of packets may be
discarded, which can lead to call quality degradation(降解). If a jitter buffer
is too large then the additional delay can lead to conversational difficulty.
Resolution
A typical jitter buffer configuration is 30mS to 50mS in size. In the case
of an adaptive jitter buffer then the maximum size may be set to 100-200mS.
Note that if the jitter buffer size exceeds(超过) 100mS then the additional delay
introduced can lead to conversational difficulty.
宾馆上网，其实用交换机就可以
不过由于两个人都是用笔记本，用无线上网比较方便，就搞个无线路由器用。
通过192.168.1.1连接到无线路由的设置页面，设置好路由器名字、密码等，用笔记本连接，能连上无线路由器，但是上不了网。
打电话问宾馆网管，答曰宾馆只允许一个端口上网，不可能多人上网，纯粹瞎扯，不理他，放下电话百度之。
最后是这么解决的：
1 先用一台笔记本直接连接到宾馆网口，ip、dns都设置成自动获取。
2 运行 ipconfig /all
看到ip地址是92.168.1.212
网关192.168.1.1
DNS 服务器 219.150.150.150
备用dns 211.161.142.10
3 把路由器连接到宾馆网口，笔记本连接到路由器上
4通过192.168.1.1连接到无线路由的设置页面
网络参数、LAN 口设置
原来是192.168.1.1，将其改为192.168.0.1，这时路由器会重启
5通过192.168.0.1连接到无线路由的设置页面
网络参数、WAN 口设置
连接类型设置为：静态 IP
IP 地址设置为 ：192.168.1.9 （2 ～ 255都可以）
网关仍然是：192.168.1.1
DNS 服务器 219.150.150.150
备用dns 211.161.142.10
保存、退出
OK，现在两个人都可以无线上网了，不用担心在宾馆里面被网线绊倒了。
流媒体是指Internet上使用流式传输 技术的连续时基媒体。当前在Internet上传输音频和视频等信息主要有两种方式：。
下载情况下，用户需要先。在视频直播等应用场合，由于生成整个媒体文件要等直播结束，也就 是用户至少要在直播结束后才能看到直播节目，所以用下载方式不能实现直播。
流式传输是指传输之前首先对多媒体进行预处理(降低质量和高效压缩)，然后使用缓存系统来保证数据连续正确地进 行传输。使用流式传输可以流媒体节目，使传输那些事先不知道或无法知道大小的媒 体数据(如网上直播、视频会议等) 成为可能。
目前，支持流媒体传输的协议主要有：实时传输协议RTP、实时传输控制协议RTCP和实 时流协议RTSP(Real-time Streaming Protocol) 等。
实时传输协议 RTP(Real-time Transport Protocol)
RTP是IETF提出的一个标准，对应的RFC文档为RFC3550（RFC1889为其过期版本）。 RFC3550不仅定义了RTP，而且定义了配套的相关协议RTCP（Real-time Transport Control Protocol，即实时传输控制协议）。
。多媒体数据块经 压缩编码处理后，先送给 RTP 封装成为 RTP 分组，再装入运输层的 UDP 用户数据报，然后再交给 IP 层。
RTP的协议层次
从应用开发者的角度看，RTP 应当是应用层的一部分。在应用的，开发者必须编 写的程序代码，然后把 RTP
for (;;) {}
8048352: eb fe jmp 8048352 <main+0xe>
8048354: 90 nop
8048355: 90 nop
8048356: 90 nop
8048357: 90 nop
8048358: 90 nop
8048359: 90 nop
804835a: 90
http://www.ibm.com/developerworks/cn/linux/l-sigdebug.html
将信号用作 Linux 调试工具
linux i386下修改, 可以应用, 能用综合的工具应用定位到行就好(不知道gdb是不是这么实现的, :) )
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <signal.h>
#include <errno.h>
#include <ucontext.h>
static void seghandler(int sn, siginfo_t * si, void *uc)
{
struct ucontext *sc = (struct ucontext *)uc;
unsigned int meip;
int i;
/*
meip = *(unsigned int *)(((struct pt_regs *)
((&(sc->uc_mcontext))->regs))->eip);
*/
printf(" Signal number = %d, Signal errno = %d\n",
si->si_signo, si->si_errno);
switch (si->si_code)
{
case 1:
printf(" SI code = %d (Address not mapped to object)\n",
si->si_code);
&nb
