dedicated bearer

Session Initiation Protocol (SIP) Basic Call Flow Examples   RFC3665

 intuitive,   done.

 

 

PDF]

TS 123 401 - V11.3.0 - LTE; GPRS enhanced for LTE

 

BASICS OF LTE bearer;

 

 

 

 

 3GPP36331 

 

 developer.

 

kamailio-devel-guide

 

DOC　many in one. 

 sip router

 

 

 

---

 

节后归来。

 

However, before starting
to tweak and adapt to your needs, we recommend that you do yourself a favour and read up on SIP. Please refer

http://sip-router.org/docbook/sip-router/branch/master/sip/sip_introduction.html

 

and more depth pdf doc can be download from  this.

http://www.iptel.org/sip

 

and official doc is 3261.

 

 

 

####### Routing Logic ########


# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {

    # per request initial checks
    route(REQINIT);

    # NAT detection
    route(NAT);

    # handle requests within SIP dialogs
    route(WITHINDLG);

    ### only initial requests (no To tag)

    # CANCEL processing
    if (is_method("CANCEL"))
    {
        if (t_check_trans())
            t_relay();
        exit;
    }

    t_check_trans();

    # authentication
    route(AUTH);

    # record routing for dialog forming requests (in case they are routed)
    # - remove preloaded route headers
    remove_hf("Route");
    if (is_method("INVITE|SUBSCRIBE"))
        record_route();

    # account only INVITEs
    if (is_method("INVITE"))
    {
        setflag(FLT_ACC); # do accounting
    }

    # dispatch requests to foreign domains
    route(SIPOUT);

    ### requests for my local domains

    # handle presence related requests
    route(PRESENCE);

    # handle registrations
    route(REGISTRAR);

    if ($rU==$null)
    {
        # request with no Username in RURI
        sl_send_reply("484","Address Incomplete");
        exit;
    }

    # dispatch destinations to PSTN
    route(PSTN);

    # user location service
    route(LOCATION);

    route(RELAY);
}

 

 

 

 

 

In order to understand NAT and  RTP proxying, you must understand what happens when a user agent registers with a SIP Registrar and when a
call is made.

 

 

 

 

Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) also known as STUN…

Application Level Gateway (ALG).

Session Border Controllers (SBC).

 

在voip-info的副标题他这样写道

A REFERENCE GUIDE TO ALL THINGS VOIP.

ALL THINGS .... 

 

今天了解到那个英国留学的海龟同事五一三天都在家写代码。。。五味杂陈

i Can not attribute this explanation to  young and  naive 

 

 

lr

  loose routing was required we cannot process the message any further  

 

 

 note that unless SER is compiled with mode=debug, this may not work on all architectures. SER will just print some basic stuff and then be quiet.

 

When we receive a REGISTER message, we immediately send a 100 Trying message back to the SIP client to stop it from retransmitting REGISTER messages. Since SER is UDP based there is no guaranteed delivery of SIP messages, so if the sender does not get a reply back quickly then it will retransmit the message.

 

 

 

---

# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722

 

 

This SER configuration handles all NAT related issues invisibly so that configuring SIP phones is a breeze. We do not use STUN because STUN generally adds another layer of complexity that can be avoided.

 

http://nil.uniza.sk/

 

NOTE:   In cases where both SIP clients are on the public Internet, then we do not proxy RTP streams since both SIP clients can directly
contact each other. This is a key to building a scaleable VoIP platform.

we don need proxy more

 

 

 

 ip addr show 

 

 

 

 

 

 

转载于:https://www.cnblogs.com/iamgoodman/p/3696690.html