[SR-Users] SDPOPS issue or append_hf

Daniel-Constantin Mierla miconda at gmail.com
Wed Aug 6 18:35:33 CEST 2014


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; '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.
>
>
>
>     _______________________________________________
>
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
>     sr-users at lists.sip-router.org  <mailto: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://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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