Changes between Version 1 and Version 2 of Using_SIP_TCP


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Timestamp:
Apr 13, 2012 6:35:05 AM (13 years ago)
Author:
bennylp
Comment:

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  • Using_SIP_TCP

    v1 v2  
    1212[[BR]] 
    1313 
    14 == Enabling TCP support == 
     14== Enabling TCP support == #enable 
    1515 
    16 First you must instantiate SIP TCP transport in your application. The TCP transport is enabled by default on ''pjsua'' so you can skip this test if you're using pjsua. 
     16TCP support must be enabled in the build by setting '''{{{PJ_HAS_TCP}}}''' to non-zero (it is enabled by default). You must then instantiate SIP TCP transport in your application, e.g.: 
     17 
     18 {{{ 
     19pjsua_transport_config  tcfg; 
     20 
     21pjsua_transport_config_default(&tcfg); 
     22status = pjsua_transport_create(PJSIP_TRANSPORT_TCP, &tcfg, &transport_id); 
     23 }}} 
    1724 
    1825 
    19 == Sending Initial Requests == 
     26[[BR]] 
    2027 
    21 PJSIP only sends the request with TCP if the destination URI contains ''";transport=tcp"'' parameter. Adding this parameter can be accomplished in two ways: 
    22  1. the most convenient way is to add a route-set entry (with proxy or outbound proxy setting) containing URI with TCP transport. This way all '''initial''' requests will be sent with TCP via the proxy, and we don't need to change the URI for the registrar and all buddies in the buddy list. Sample code: 
     28 
     29== Sending Initial Requests == #initial 
     30 
     31According to SIP spec, a request is sent to the URI in the ''Route'' header if it is present, or to the request URI if not. PJSIP only sends the request with TCP if the destination URI contains ''";transport=tcp"'' parameter. Hence sending a request with TCP can be accomplished in two ways: 
     32 
     33 1. the most convenient way is to add a route-set entry (with proxy or outbound proxy setting in the account config) containing URI with TCP transport. This way all '''initial''' requests will be sent with TCP via the proxy, and we don't need to change the URI for the registrar and all buddies in the buddy list. Sample code: 
    2334 {{{ 
    2435pjsua_acc_config acc_cfg; 
     
    2839          
    2940 }}} 
    30  2. if you don't want to configure route set entry, then you must add ''";transport=tcp"'' parameter to all outgoing URIs (the registrar URI, the buddy URI, the target URI when making call, the target URI when sending MESSAGE, etc.). For example, to make outgoing call with TCP: 
     41 2. if you don't want to configure route set entry, then you must add ''";transport=tcp"'' parameter to all outgoing URIs (the registrar URI, the buddy URI, the target URI when making calls, the target URI when sending MESSAGE, etc.). For example, to make outgoing call with TCP: 
    3142 {{{ 
    3243pj_str_t dst = pj_str("sip:alice@example.net;transport=tcp"); 
     
    3546 }}} 
    3647 
    37 == Contact Header == 
     48[[BR]] 
     49 
     50 
     51== Contact Header == #contact 
    3852 
    3953With PJSUA-LIB, when making or receiving calls with TCP, the local Contact header will automatically be adjusted to use the TCP transport. 
    4054 
    41 == Subsequent Requests == 
    4255 
    43 Subsequent requests means subsequent request that is sent within the call (dialog), for example UPDATE, BYE, re-INVITE to hold the call, and so on. 
     56[[BR]] 
     57 
     58== Subsequent Requests == #nextreq 
     59 
     60Subsequent requests means subsequent request that is sent within the call (dialog), for example UPDATE, BYE, re-INVITE to hold the call, and so on. Subsequent requests within a dialog will be sent to the URI that is found in the top-most ''Route'' header which was built from the ''Record-Route'' header in the response that established the dialog (it could be the 18x or 200/OK response), or if there's no ''Route''/''Record-Route'', the URI in the ''Contact'' header of that response. 
     61 
     62It could be the case that the initial request is sent with TCP, but the subsequent ones are with UDP. In this case, check the URI in the ''Route'' or ''Record-Route'' or ''Contact'' header of the 18x or 2xx response that is sent by the remote party. Chances are this header lacks the ''";transport=tcp"'' parameter in the URI; in this case, you can either configure the other end to use TCP, or configure your proxy to ''record-route'' (i.e. to force itself to be within the request path of the call). 
    4463 
    4564 
    46 == Additional Info about Registration == 
     65[[BR]] 
     66 
     67== Automatic Switch to TCP if Request is Larger than 1300 bytes == #switch 
     68 
     69According to [http://tools.ietf.org/html/rfc3261#section-18.1.1 RFC 3261 section 18.1.1]:  
     70 
     71  "If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such  as TCP." 
     72 
     73By this rule, PJSIP will automatically send the request with TCP if the request is larger than 1300 bytes. The switching is done on request by request basis, i.e. if an initial INVITE is originally meant to use UDP but end up being sent with TCP because of this rule, then only that initial INVITE is sent with TCP; subsequent requests will use UDP, unless of course if it's larger than 1300 bytes. In particular, the Contact header stays the same. Only the Via header is changed to TCP. 
     74 
     75It could be the case that the initial INVITE is sent with UDP, and once the request is challenged with 401 or 407, the size grows larger than 1300 bytes due to the addition of ''Authorization'' or ''Proxy-Authorization'' header. In this case, the request retry will be sent with TCP. 
     76 
     77In case TCP transport is not instantiated, you will see error similar to this: 
     78 
     79  ''"Temporary failure in sending Request msg INVITE/cseq=15228 (tdta02EB0530), will try next server. Err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))'' 
     80 
     81As the error says, the error is not permanent, as PJSIP will send the request anyway with UDP. 
     82 
     83This TCP switching feature can be disabled as follows: 
     84 * at run-time by setting {{{pjsip_cfg()->endpt.disable_tcp_switch}}} to PJ_TRUE. 
     85 * at-compile time by setting {{{PJSIP_DONT_SWITCH_TO_TCP}}} to non-zero 
     86 
     87You can also tweak the 1300 threshold by setting {{{PJSIP_UDP_SIZE_THRESHOLD}}} to the appropriate value. 
     88 
     89 
     90[[BR]] 
     91 
     92== Additional Info about Registration == #reg 
    4793 
    4894The client registration session also will keep the TCP connection active throughout the registration session, and server may send inbound requests using this TCP connection if it wants to.  
     95 
     96 
     97[[BR]] 
     98 
    4999 
    50100