[SR-Users] SDPOPS issue or append_hf

Daniel-Constantin Mierla miconda at gmail.com
Wed Aug 6 16:42:12 CEST 2014


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
> *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.
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140806/2c69f260/attachment.html>


More information about the sr-users mailing list