18 May
2015
18 May
'15
2:26 p.m.
Hi
Did you use msg_apply_changes() before relaying the INVITE
?http://kamailio.org/docs/modules/4.1.x/modules/textopsx.html#textopsx.f.msg_apply_changes
Regards,Dragos
From: José Seabra <joseseabra4(a)gmail.com>
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>
Sent: Monday, May 18, 2015 12:31 PM
Subject: [SR-Users] Function sdp_remove_codecs_by_id seems to be not working
Hello,
I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove
some codecs in INVITE request before send it to freeswitch, but the function doesn't
remove the codec, and it doesn't give any error message.
I'm using this function in request route.
Kamailio version is 4.2.2.
INVITE that kamailio receives from phone:
INVITE sip:401@teste.demo.pt;user=phone SIP/2.0Record-Route:
<sip:10.0.20.102:5062;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes>Record-Route:
<sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes>Via: SIP/2.0/UDP
10.0.20.102:5062;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0Via: SIP/2.0/UDP
192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060From:
"301" <sip:301@teste.demo.pt>;tag=oztyflbzbxTo:
<sip:401@teste.demo.pt;user=phone>Call-ID: 3c3a58a25d63-ghfc5xdg1sn0CSeq: 1
INVITEMax-Forwards: 69Contact:
<sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1X-Serialnumber:
000413262FA0P-Key-Flags: resolution="31x13", keys="4"User-Agent:
snom370/8.4.35Accept: application/sdpAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS,
NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATEAllow-Events: talk, hold, refer,
call-infoSupported: timer, 100rel, replaces, from-changeCall-Info:
<sip:teste.demo.pt>;appearance-index=1Session-Expires: 3600;refresher=uasMin-SE:
90Content-Type: application/sdpContent-Length: 391v=0o=root 24935823 24935823 IN IP4
192.168.10.147s=callc=IN IP4 192.168.10.147t=0 0m=audio 19410 RTP/AVP 0 8 9 99 3 18 4
101a=rtpmap:0 PCMU/8000.a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:99
G726-32/8000a=rtpmap:3 GSM/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:4
G723/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
INVITE that kamailio send to freeswitch after execute
sdp_remove_codecs_by_id("18"):
INVITE sip:401@teste.demo.pt;user=phone SIP/2.0.Record-Route:
<sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>.Record-Route:
<sip:10.0.20.102:5062;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.Record-Route:
<sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>.Via: SIP/2.0/UDP
10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0.Via: SIP/2.0/UDP
10.0.20.102:5062;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0.Via: SIP/2.0/UDP
192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060.From:
"301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9.To:
<sip:401@teste.demo.pt;user=phone>.Call-ID: 3c3a7c84e065-pr2hm0uk9yfz.CSeq: 2
INVITE.Max-Forwards: 68.Contact:
<sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1.X-Serialnumber:
000413262FA0.P-Key-Flags: resolution="31x13", keys="4".User-Agent:
snom370/8.4.35.Accept: application/sdp.Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS,
NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE.Allow-Events: talk, hold, refer,
call-info.Supported: timer, 100rel, replaces, from-change.Call-Info:
<sip:teste.itcenter.com.pt>;appearance-index=1.Session-Expires:
3600;refresher=uas.Min-SE: 90.Content-Type: application/sdp.Content-Length: 403..
v=0.o=root 228603317 228603317 IN IP4 100.64.250.4.s=call.c=IN IP4 100.64.250.4.t=0
0.m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101.a=rtpmap:0 PCMU/8000.a=rtpmap:8
PCMA/8000.a=rtpmap:9 G722/8000.a=rtpmap:99 G726-32/8000.a=rtpmap:3 GSM/8000.a=rtpmap:18
G729/8000.a=fmtp:18 annexb=no.a=rtpmap:4 G723/8000.a=rtpmap:101
telephone-event/8000.a=fmtp:101 0-16.a=ptime:20.a=sendrecv.a=rtcp:49405.
SDP body has no changes related with codecs.
Anyone call help please.
Thank youBRJosé Seabra
--
CumprimentosJosé Seabra
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