[OpenSIPS-Users] sipmsgops.so / codec_delete_except_re
Bogdan-Andrei Iancu
bogdan at opensips.org
Fri Sep 27 06:51:04 EDT 2019
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: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/
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