I will try to look at this. It’s a bit tricky as the call-flow is not the most easiest.
It’s strange because there is nothing to delete in that case because the list of codecs is already okay.
Regards,
Igor.
De : Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Envoyé : mercredi 6 août 2014 18:36
À : Igor Potjevlesch; 'Kamailio (SER) - Users Mailing List'
Objet : Re: [SR-Users] SDPOPS issue or append_hf
It looks related to how changes are done to a sip message. rtpproxy is working on incoming message as well as sdpops. Practically, rtpproxy adds a new line at the end of the incoming sdp. sdopos deletes from old sdp, resulting in empty lines inside the sdp.
Can you do the sdpops operation before record_route() and after it call msg_apply_changes() from textopsx module?
Cheers,
Daniel
On 06/08/14 17:48, Igor Potjevlesch wrote:
It’s really linked to the initial SDP. If I have only one codec, for example G711u (plus telephone-event), and I just keep G711u, a blank line is inserted.
If I keep G711u + telephone-event, everything is working fine.
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com]
Envoyé : mercredi 6 août 2014 17:25
À : miconda@gmail.com; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] SDPOPS issue or append_hf
Hello Daniel,
I got a feedback from the telco in the meantime. He told me that the issue is the blank line between “rtpmap:8..” and “nortpproxy”.
This parameter is supported. I have successful calls with “nortpproxy=yes”.
I don’t know why sdp_keep_codecs_by_name inserts a blank line here.
Regards,
Igor.
De : sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Daniel-Constantin Mierla
Envoyé : mercredi 6 août 2014 16:42
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] SDPOPS issue or append_hf
Hello,
the problem here is with rtpproxy marker -- can you try with the parameter set to empty string?
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856
Cheers,
DanielOn 06/08/14 12:23, Igor Potjevlesch wrote:
Hello,
To be sure that the issue is not coming from append_hf, I add (…,”Call-ID”). The PAI is now inserted after the Call-ID.
But, the issue remains:
Content-Type: application/sdp
Content-Length: 169
v=0
o=UserA 1153072414 140968390 IN IP4 A.B.C.D
s=Session SDP
c=IN IP4 A.B.C.D
t=0 0
m=audio 60412 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
This SDP is dropped. Someone see something missing or wrong in the SDP parts?
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com]
Envoyé : mercredi 6 août 2014 11:57
À : sr-users@lists.sip-router.org
Objet : SDPOPS issue or append_hf
Hello,
I have an issue with the module SDPOPS while using “sdp_keep_codecs_by_name”.
If the calling party sends only one codec description like:
Content-Type: application/sdp
Content-Length: 202
v=0
o=UserA 2966746938 1790378070 IN IP4 10.141.0.21
s=Session SDP
c=IN IP4 10.141.0.21
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is:
Content-Type: application/sdp
Content-Length: 170
P-Asserted-Identity: "+0123456789" <sip:+0123456789@sip.tld>
v=0
o=UserA 2485672881 3000549892 IN IP4 a.b.c.d
s=Session SDP
c=IN IP4 a.b.c.d
t=0 0
m=audio 40330 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=nortpproxy:yes
If I open the capture in Wireshark, the PAI is not in the SDP part, and the end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18 bytes)”.
I don’t understand why the PAI is inserted within the SDP part. Adding the PAI is done after “sdp_keep_codecs_by_name”:
if (!is_present_hf("P-Asserted-Identity")) {
$var(pai) = $(fU{re.subst,/^0/+33/g});
append_hf("P-Asserted-Identity: \"$var(pai)\" <sip:$var(pai)@$fd>\r\n");
}
I guess that this cause my INVITE being dropped by 488 Media Not Acceptable Here.
Regards,
Igor.
_______________________________________________SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/micondaNext Kamailio Advanced Trainings 2014 - http://www.asipto.comSep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA