Hi Alexey,
The logs are quite self-explanatory:
DBG:sipmsgops:do_for_all_streams: Message has no SDP
Where in the script do you call the codec_delete_except_re() function and for
which SIP message ? Please the blow line just before calling
codec_delete_except_re():
xlog("--> $rm/$rr, $rb\n");
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 9/13/19 3:59 PM, Alexey Kazantsev via Users wrote:
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
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
