sip sdp

find dialplan.The system searches for the dialplan corresponding to the called sip settings and loads it To the memory,Call status conversion: cs_routing-> cs_execut. The state machine calls switch_core_standard_on_execute and executes the loaded dialplan one by one according to the rule. This is the embodiment of programmable SoftSwitch, flexibly control the service provided to a call as needed. When the bridge action is executed, the system calls u

process to match the FreeSWITCH default codec, to find the support type's specific rate information from the loaded module Sofia_handle_sip_i_invite=> Sofia_glue_tech_prepare_codecs=> switch_loadable_module_get_codecs_sorted negotiated to be called SDP Sofia_glue_do_invite=> Sofia_glue_tech_prepare_codecs=> switch_loadable_module_get_codecs_sorted Ocodec +codec_string generates m headers and sends invite containing the

Call persistence process and call ProcessCall persistence FlowchartCall persistence signaling: 1) when user 1000 sends the re-INVITE message to IPBX, the call between 1000 and 1002 needs to be maintained. The Re-INVITE message carries c = IN IP4 0.0.0.0 in sdp, indicating that the re-INVITE message is a call persistence message. INVITE sip: 1002@192.168.50.22: 5060SIP/2.0 Via:

Entry Point of NGN access control security -- Diameter protocol and its application in the SIP network environment Xie Wei I. Introduction The diameter series protocol is a new generation of AAA technology, which is gaining more and more attention due to its powerful scalability and security assurance. In international standards organizations such as ITU, 3GPP and PP2, DIAM-ETER protocols have been officially used as the preferred AAA protocol for fut

is used to control the connection, establishment, and release of the Media/control flow on the media gateway and MGCP terminal. Session Initiation Protocol (SIP) Session Initiation Protocol (SIP) is an application layer control protocol proposed by IETF for multimedia communication over IP networks. Based on the Internet protocol HTTP), follow the Internet design principles, and work in peer-to-peer mode.

Document directory RTP Stacks (mainly open source C/C ++ stacks) SIP Stacks RTP Applications SIP Phones (SIP User Agents) SIP Test Utility SIP Applications (Proxy, Location Server) Sip Express Router (ser) Ser Media Serv

VoIP bookmarks from Klaus darilion Below you will find descriptions and links to sip and RTP stacks, applications, test utilities, SIP proxies, SIP pbxs and stun server and clients. most of them are open source :-), but not all of them If you have any comments please feel free to contact me: --> Klaus. darilion at pernau. at There are also other VoIP related por

Document directory Jain proposal SIP, ISUP, call control system, and Jain Interface Application of Jain APIs to Mobile Networks Mobile Station No-wire access to network (RAN) Network and Enterprise Internal capacity and service End-to-end structure Jain API for call control and Wireless Networks Face-to-Face Jain API of the integrated network connects the business agility, network convergence, and security network to the telephone and

with the peer-to-peer approach, and some scenarios may be no matter what NAT type is Media-relay way, This method increases end-to-end latency and packet loss probability. Stun and turn are client/server protocols, the white is the client to the server to their own public network address and port, and then placed in their own invite request SDP message body and the invite of the "OK SDP" message body. Most

MessagesSofia.c:sofia_handle_sip_i_state...Mod_dialplan_xml.c:dialplan_hunt (Ring_ready)Resolve the dialing scheme, execute the LUA script, and set the channel variables....Send 180 eventsData Sheet involved:Sip_registrations, Ip_dialogs3, send invite to B extension, specific as follows:Handling Nua_r_invite MessagesSofia_handle_sip_r_invite = originate_on_routingData Sheet involved:Sip_dialogs4, B answer, the two sides of the call, specifically as follows:Sofia_receive_message:switch_message_i

so that the user or UA can select the most appropriate address to forward the request. The MIME type is not defined for the message body without this response.These address selections should also be listed in the Contact Header field (section 20.10). Unlike the Http,sip answer, you can include multiple contact header fields or a Contact header field that has an address list. UA can use the Contact header field to automatically forward or ask the user