Hi Bogdan, in fact, I'm just playing with this and have no real need (at least right now). But I'm curious why the function is not working properly for me.
Script debugging shows smth strange about SDP presence: DBG:sipmsgops:create_codec_lumps: creating 1 streams DBG:sipmsgops:get_associated_lump: Have 1 lumps DBG:sipmsgops:get_associated_lump: have lump at 619 want at 619 DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_sdp: message body has length zero DBG:sipmsgops:do_for_all_streams: Message has no SDP DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_sdp: message body has length zero DBG:sipmsgops:do_for_all_streams: Message has no SDP But the SIP debug seems to be OK. This is the INVITE from Linphone to OpenSIPS: 2019/09/13 17:50:18.815477 195.209.116.18:5060 -> 185.212.148.195:5060 INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 195.209.116.18:5060;rport;branch=z9hG4bK319001858 From: <sip:[email protected]>;tag=1401125272 To: <sip:[email protected]> Call-ID: 910262535 CSeq: 20 INVITE Contact: <sip:[email protected]:5060> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Phone call Content-Length: 237 v=0 o=alexey 2285 1260 IN IP4 195.209.116.18 s=Talk c=IN IP4 195.209.116.18 t=0 0 m=audio 7078 RTP/AVP 0 8 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 And this is the INVITE coming from OpenSIPS to another UAC (here I'd like any codec except pcm* be removed from SDP): 2019/09/13 17:50:18.823409 185.212.148.195:5060 -> 195.209.116.4:5061 INVITE sip:[email protected]:5061 SIP/2.0 Record-Route: <sip:185.212.148.195;lr> Via: SIP/2.0/UDP 185.212.148.195:5060;branch=z9hG4bK4291.21f47e33.0 Via: SIP/2.0/UDP 195.209.116.18:5060;received=195.209.116.18;rport=5060;branch=z9hG4bK319001858 From: <sip:[email protected]>;tag=1401125272 To: <sip:[email protected]> Call-ID: 910262535 CSeq: 20 INVITE Contact: <sip:[email protected]:5060> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Phone call Content-Length: 266 v=0 o=alexey 2285 1260 IN IP4 185.212.148.195 s=Talk c=IN IP4 185.212.148.195 t=0 0 m=audio 30184 RTP/AVP 0 8 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendrecv a=rtcp:30185 ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
