6 Aug
2014
6 Aug
'14
11:48 a.m.
Its 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(a)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 dont know why sdp_keep_codecs_by_name inserts a blank line here.
Regards,
Igor.
De : sr-users-bounces(a)lists.sip-router.org
<mailto: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,
Daniel
On 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(a)lists.sip-router.org <mailto: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 dont 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 <sip:$var%28pai%29@$fd> >\r\n");
}
I guess that this cause my INVITE being dropped by 488 Media Not Acceptable
Here.
Regards,
Igor.
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--
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