[OpenSIPS-Users] sipmsgops.so / codec_delete_except_re

[Alexey Kazantsev]

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:lexus2 at alexeyka.zantsev.com SIP/2.0
   Via: SIP/2.0/UDP 195.209.116.18:5060;rport;branch=z9hG4bK319001858
   From: <sip:lexus at alexeyka.zantsev.com>;tag=1401125272
   To: <sip:lexus2 at alexeyka.zantsev.com>
   Call-ID: 910262535
   CSeq: 20 INVITE
   Contact: <sip:lexus at 195.209.116.18:5060>
   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:lexus2 at 195.209.116.4:5061 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:lexus at alexeyka.zantsev.com>;tag=1401125272
   To: <sip:lexus2 at alexeyka.zantsev.com>
   Call-ID: 910262535
   CSeq: 20 INVITE
   Contact: <sip:lexus at 195.209.116.18:5060>
   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/