[Kamailio-Users] Kamailio RTP Proxy issues

[General Lee]

Thank you Alex.

If the rtpproxy_offer/answer functions do not work, do you have any ideas on
how I should route an INVITE without SDP and ACK with SDP using
force_rtp_proxy similar to the example below?


route {
...
    if (is_method("INVITE")) {
        if (has_sdp()) {
            if (rtpproxy_offer())
                t_on_reply("1");
        } else {
            t_on_reply("2");
        }
    }
    if (is_method("ACK") && has_sdp())
        rtpproxy_answer();
...
}

onreply_route[1]
{
...
    if (has_sdp())
        rtpproxy_answer();
...
}

onreply_route[2]
{
...
    if (has_sdp())
        rtpproxy_offer();
...
}



Alex Balashov wrote:
> 
> General Lee,
> 
> We just had a thread about this, initiated by me and Joe Hart.  We 
> came to the conclusion that the rtpproxy_offer/answer functions don't 
> actually work, but force_rtp_proxy() does, and that use of flags is 
> required:
> 
> http://lists.kamailio.org/pipermail/users/2009-October/024934.html
> 
> -- Alex
> 
> General Lee wrote:
> 
>> Hi Klaus.
>> 
>> I use the rtpproxy_offer + answer functions without any flags. I've
>> listed
>> parts of my code below.
>> 
>> route[2]
>> {
>> 	if (is_method("BYE|CANCEL"))
>> 	{
>> 		unforce_rtp_proxy();
>> 	} else if (is_method("INVITE")) {
>> 		if(has_body("application/sdp")){
>> 			if(rtpproxy_offer())
>> 				t_on_reply("1");
>> 		}else{
>> 			t_on_reply("2");		#this will handle the initial INVITE that has no SDP
>> 		}
>> 	}
>> 	if(is_method("ACK") && has_body("application/sdp")){
>> 			rtpproxy_answer();	
>> }
>> }
>> 
>> 
>> 
>> 
>> onreply_route[1]
>> {
>> 
>> 	if (has_body("application/sdp"))
>> 		rtpproxy_answer();
>> 
>> 	if (isbflagset(6))
>> 	{
>> 		search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nty=yes');
>> 		search_append('m:.<sip:[^>[:cntrl:]]*', ';nty=yes');
>> 		if(cmp_str("MY_IP","$si"))
>> 		else {
>> 			fix_nated_contact();
>> 		}
>> 	}
>> 	exit;
>> }
>> 
>> onreply_route[2]
>> {
>> 	if (has_body("application/sdp"))
>> 		rtpproxy_offer();
>> 		
>> 	if (isbflagset(6))
>> 	{
>> 		search_append('Contact:.*sip:[:cntrl:]]*', ';nty=yes');
>> 		search_append('m:.<sip:[^>[:cntrl:]]*', ';nty=yes');
>> 		if(cmp_str("MY_IP","$si"))
>> 		else {
>> 			fix_nated_contact();
>> 		}
>> 	}
>> 	exit;
>> }
>> 
>> 
>> 
>> 
>> Klaus Darilion-2 wrote:
>>> Do you use force_rtpproxy() or the ...offer() and ...answer() functions?
>>>
>>> Do you use any flags when call the functions?
>>>
>>> klaus
>>>
>>> General Lee schrieb:
>>>> Hi,
>>>>
>>>> I am currently integrating an H323 to SIP gateway with Kamailio and
>>>> trying
>>>> to route all calls through RTP Proxy. 
>>>>
>>>> I have a problem where RTP Proxy treats both the incoming H323 gateway
>>>> call
>>>> (Caller) and outgoing SIP call (Callee) as the 'caller' in the RTP
>>>> Proxy
>>>> syslog. RTP Proxy doesn't assign a 'callee' therefore not able to setup
>>>> a
>>>> call (See syslog below). I have configured Kamailio to accept the 'ACK'
>>>> with
>>>> SDP and this is working correctly.
>>>>
>>>> When I enable H323 Fast connect and the SDP is included in the INVITE,
>>>> the
>>>> call connects correctly and is routed through RTP Proxy. I feel the
>>>> problem
>>>> is related to RTP Proxy receiving an INVITE from the H323-SIP gateway
>>>> without SDP.
>>>>
>>>>  Can anyone explain why RTP Proxy treats both the incoming H323 Gateway
>>>> call
>>>> and outgoing SIP call as the 'caller' in the RTP Proxy syslog. How can
>>>> I
>>>> make RTP Proxy treat the incoming H323 call as the 'callee'? 
>>>>
>>>> Thanks,
>>>>
>>>>
>>>>
>>>>
>>>> More information below
>>>>
>>>> *******************************************************************************
>>>>
>>>> My H323 endpoints use H323 Slow Connect, so when the H323-SIP Gateway
>>>> delivers the 'INVITE' to Kamailio there is no SDP included in the
>>>> INVITE.
>>>> I
>>>> added a 'onreply_route' to the Kamailio configuration file which
>>>> handles
>>>> the
>>>> 'ACK' with SDP which is working correctly.
>>>>
>>>> All of my SIP calls (Signalling + Media) are forced though RTP Proxy
>>>> and
>>>> I
>>>> would like to force all H323-SIP Gateway calls through RTP Proxy.
>>>>
>>>> When placing a call from my H323 endpoint to my SIP UA, the RTP Proxy
>>>> syslog
>>>> records the incoming and outgoing call, however RTP Proxy states that
>>>> the
>>>> ‘callee’ is actually the 'caller'. The RTP Proxy syslog also states
>>>> that
>>>> the
>>>> caller is the caller so there is no 'callee' (see below).  In the
>>>> syslogs
>>>> both the ‘callee’ and ‘caller’ are recognised as the ‘caller’ so RTP
>>>> Proxy
>>>> has no callee to send the traffic back to.
>>>>
>>>> When the INVITE is received the ‘callee’ is populated in the syslogs as
>>>> the
>>>> ‘caller’. The H323 Gateway call is not recorded until after the ‘ACK’ 
>>>> with
>>>> SDP is received from the gateway.
>>>>
>>>> Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 72.19.211.106:49620 (Should be callee)
>>>> Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 72.19.211.106:49622  (Should be callee)
>>>>
>>>> Then after the ‘ACK’ is received from the Gateway, the H323 call is
>>>> mentioned in the syslog as well as the other caller who is supposed to
>>>> be
>>>> the callee. 
>>>>
>>>> Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 69.72.11.51:10204
>>>> Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 69.72.11.51:10214
>>>>
>>>> Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 72.19.211.106:49620
>>>> Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
>>>> caller's
>>>> address with 72.19.211.106:49622
>>>>
>>>>   
>>> _______________________________________________
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>>> Users at lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>
>> 
> 
> 
> -- 
> Alex Balashov - Principal
> Evariste Systems
> Web     : http://www.evaristesys.com/
> Tel     : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
> 
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