[SR-Users] SDPOPS issue or append_hf

Thu Aug 7 14:02:44 CEST 2014

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 at 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 at gmail.com] 
Envoyé : mercredi 6 août 2014 17:25
À : miconda at gmail.com <mailto:miconda at 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 at lists.sip-router.org
<mailto:sr-users-bounces at lists.sip-router.org>
[mailto:sr-users-bounces at 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 at gmail.com] 
Envoyé : mercredi 6 août 2014 11:57
À : sr-users at lists.sip-router.org <mailto:sr-users at 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 at 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)@$fd <sip:$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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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/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

-- 
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
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