한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

    
페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


http://www.opensips.org/html/docs/modules/devel/tm.html



TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly in terms of memory and CPU, is some services inherently need state. For example, transaction-based accounting (module acc) needs to process transaction state as opposed to individual messages, and any kinds of forking must be implemented statefully. Other use of stateful processing is it trading CPU caused by retransmission processing for memory. That makes however only sense if CPUconsumption per request is huge. For example, if you want to avoid costly DNS resolution for every retransmission of a request to an unresolvable destination, use stateful mode. Then, only the initial message burdens server by DNS queries, subsequent retransmissions will be dropped and will not result in more processes blocked by DNS resolution. The price is more memory consumption and higher processing latency.

From user's perspective, the major function is t_relay(). It setup transaction state, absorb retransmissions from upstream, generate downstream retransmissions and correlate replies to requests.

In general, if TM is used, it copies clones of received SIP messages in shared memory. That costs the memory and also CPUtime (memcpys, lookups, shmem locks, etc.) Note that non-TM functions operate over the received message in private memory, that means that any core operations will have no effect on statefully processed messages after creating the transactional state. For example, calling record_route after t_relay is pretty useless, as the RR is added to privately held message whereas its TMclone is being forwarded.

TM is quite big and uneasy to program--lot of mutexes, shared memory access, malloc and free, timers--you really need to be careful when you do anything. To simplify TM programming, there is the instrument of callbacks. The callback mechanisms allow programmers to register their functions to specific event. See t_hooks.h for a list of possible events.

Other things programmers may want to know is UAC--it is a very simplistic code which allows you to generate your own transactions. Particularly useful for things like NOTIFYs or IM gateways. The UAC takes care of all the transaction machinery: retransmissions , FR timeouts, forking, etc. See t_uac prototype in uac.h for more details. Who wants to see the transaction result may register for a callback.

조회 수 :
50938
등록일 :
2014.10.04
16:58:21 (*.160.88.200)
엮인글 :
http://webs.co.kr/index.php?document_srl=241853&act=trackback&key=009
게시글 주소 :
http://webs.co.kr/index.php?document_srl=241853
List of Articles
번호 제목 글쓴이 조회 수 추천 수 날짜
22 Open Source VOIP applications, both clients and servers. admin 53059   2013-11-20
 
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 62323   2013-09-11
 
20 My new toy: Bluebox-ng admin 100295   2013-04-06
 
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 113404   2013-04-06
 
18 Asterisk Installation Asterisk Realtime configuration admin 52616   2013-04-06
 
17 The SIP Router Project admin 52260   2013-04-06
 
16 Kamailio :: A Quick Introduction admin 91210   2013-04-06
 
15 Welcome to the Smartvox Knowledgebase admin 114321   2013-04-06
 
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 61105   2013-04-06
 
13 OpenSIPS vs Asterisk admin 245870   2013-04-06
 
12 OpenSER_from_an_asterisk_POV file admin 44886   2013-04-06
 
11 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 252069   2013-03-31
 
10 rfc5766-turn-server admin 47028   2013-03-21
 
9 OpenSIPS Kick Start‎: VIDEO admin 45790   2013-02-20
 
8 OPENSIP Training VIDEO admin 44153   2013-02-20
 
7 What is new in 1.8.0 opensip admin 281014   2012-05-21
 
6 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 170372   2012-01-06
 
5 SIP 트래픽 생성 테스트 툴 admin 151870   2011-12-23
 
4 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 165034   2011-12-23
 
3 the OpenSIPS Project OpenSIP admin 55074   2011-12-14
 
2 OpenH323 Gatekeeper - The GNU Gatekeeper admin 60970   2011-12-14
 
1 The FreeRADIUS Project admin 50820   2011-12-14