[Kamailio-Users] Re move Body Part

[Daniel-Constantin Mierla]

Hello,

you should remove end of lines as well.

Cheers,
Daniel

On 06/03/2009 11:20 AM, karhu wrote:
> Hello,
>
> I'm still on removing SDP parameters from the SIP Invite method.
> So here is the initial SDP from caller : 
>
> v=0
> o=- 5 2 IN IP4 10.10.10.3
> s=CounterPath X-Lite 3.0
> c=IN IP4 10.10.10.3
> t=0 0
> m=audio 38028 RTP/AVP 107 119 0 98 8 3 101
> a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028
> a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:119 BV32-FEC/16000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> m=video 17166 RTP/AVP 115 34
> a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166
> a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166
> a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960
> a=fmtp:34 QCIF=1 MAXBR=1960
> a=rtpmap:115 H263-1998/90000
> a=rtpmap:34 H263/90000
> a=sendrecv
>
> I just want to force the use of payload 98 on the audio part, so in
> kamailio.cfg i added that : 
>
> ##### INVITE ######
> 	if (is_method("INVITE")){
> 		route(4);	
> 		record_route();
>
> #		route(10);
> 		route(20);
>                 route(21);
> 	}
>
> ######### Route Rtpproxy Bridge Mode for INVITE methods ########### 
> route[4]
> {
> 		if (lookup("location_internal")) {
> 			if (dst_ip == 10.10.10.2)
> 				if (force_rtp_proxy("FAII"))
> 					xlog("L_INFO","appel_rtpproxyFAII");
> 					t_on_reply("1");
> 			if (dst_ip == 192.168.0.2)
> 				if (force_rtp_proxy("FAEI"))
> 					xlog("L_INFO","appel_rtpproxyFAEI");
> 					t_on_reply("1");
> 		} else if (lookup("location_external")) {
> 			if (dst_ip == 10.10.10.2)
> 				if (force_rtp_proxy("FAIE"))
> 					xlog("L_INFO","appel_rtpproxyFAIE");
> 					t_on_reply("1");
> 			if (dst_ip == 192.168.0.2)
> 				if (force_rtp_proxy("FAEE"))
> 					xlog("L_INFO","appel_rtpproxyFAEE");
> 					t_on_reply("1");
> 		} else {
> 			sl_send_reply("403", "Call cannot be served here");
> 			exit;
> 			};
> }
>
> route[20]
> {
> 	if(has_body("application/sdp") && search_body("m=audio"))
> 		xlog("L_INFO", "application/sdp_present_avec_audio");
> 		subst_body('/AVP (.*) /AVP 98 /');
> }
>
> route[21]
> {
> 	if(has_body("application/sdp") && search_body("a=rtpmap:[0-9]+"))
> 		xlog("L_INFO", "application/sdp_present_avec_description_audio");
> 		subst_body('/a=rtpmap:107 BV32\/16000//');
> 		subst_body('/a=rtpmap:119 BV32-FEC\/16000//');
> #		replace_body("a=rtpmap:107 BV32/16000","");
> #		subst_body('/a=rtpmap:[0-9]+ (.*)//');
> }
>
> and i get from wireshark capture the following sdp : 
>
> v=0
> o=- 5 2 IN IP4 10.10.10.3
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.0.2
> t=0 0
> m=audio 35044 RTP/AVP 98 101
> a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028
> a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028
> a=fmtp:101 0-15
>
>
> a=rtpmap:98 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> m=video 35046 RTP/AVP 115 34
> a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166
> a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166
> a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960
> a=fmtp:34 QCIF=1 MAXBR=1960
> a=rtpmap:115 H263-1998/90000
> a=rtpmap:34 H263/90000
> a=sendrecv
> a=nortpproxy:yes
>
> First, the 98 payload is well forced.  
> The problem is i get "Call failed: Not acceptable Here" on the caller sip
> phone. Obviously the callee phone doesn't ring.. It seems that the blanck
> lines in the sdp message shouldn't prevent the callee phone to ring.
> Do you have any ideas ? or configuration modifications ? i am stick on that
> problem.
>
> Cheers, 
>
> Karhu
>   

-- 
Daniel-Constantin Mierla
http://www.asipto.com/