[OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

Bogdan-Andrei Iancu bogdan at opensips.org
Mon Aug 13 19:51:09 CEST 2012


Hi Duane,

That is really strange. Logically speaking, a change in SDP should have 
no impact on receiving (or not) a message - the SDP is just payload and 
is not even parsed by opensips (unless you ask it from script by using 
rtpproxy/mediaproxy/sipmsgops modules). Anyhow, the request should make 
it to the main route.

I rather suspect that something else is filtering your SIP traffic, 
dropping the INVITEs in the first format :-/ .

Regarding the second issue:

1) the angle brackets are not mandatory by RFC - they should be used 
only if you have URI params or a complex display name (which is not your 
case here)

2) assuming that the phone does not like TO - it should reply with a 400 
Bad request, not a 404 - a 404 represents a routing indication and 
routing is done based RURI not TO.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/09/2012 06:13 AM, Duane Larson wrote:
> I changed the following in the ctd.sh script
> Changed the default of
> "`printf "v=0\r\no=click-to-dial 0 0 IN IP4 
> 0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0 
> 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0 
> PCMU/8000\r\n"`
> To
> "`printf "v=0\r\no=click2dial 0 0 IN IP4 50.XX.XX.156\r\ns=click2dial 
> call\r\nc=IN IP4 173.XX.XX.111\r\nt=0 0\r\nm=audio 12790 RTP/AVP 0 8 
> 18 3 4 97 98\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:18 
> G729/8000\r\na=rtpmap:97 ilbc/8000\r\na=rtpmap:98 speex/8000\r\n"`
> And now it is making it into the OpenSIPS/SBC's main route.  Not sure why.
> I noticed another issue now.  My snom phone is receiving the INVITE 
> but it is replying with a "404 Not Found" error.  (If I test with a 
> Jitsi client I don't have the 404 issue)
> This shouldn't happen since the TO header is the correct  SIP URI.  
> The only thing that can be wrong is that the To: URI is not in <>
> I think the TM MI function t_uac_dlg isn't placing the <> around the 
> TO: header URI.  Reading the RFC I am not 100% sure if the <> are 
> required.
> U 2012/08/08 22:09:13.756976 192.168.88.1:5060 
> <http://192.168.88.1:5060> -> 192.168.88.13:3072 
> <http://192.168.88.13:3072>
> INVITE sip:9016XX6XX4 at 192.168.88.13:3072 
> <http://sip:9016XX6XX4@192.168.88.13:3072> SIP/2.0.
> Max-Forwards: 10.
> Record-Route: <sip:192.168.88.1;r2=on;lr>.
> Record-Route: <sip:99.XX.XX.161;r2=on;lr>.
> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
> *To: sip:9016XX6XX4 at irck.com <mailto:sip%3A9016XX6XX4 at irck.com>.*
> From: <sip:controller at ae.com 
> <mailto:sip%3Acontroller at ae.com>>;tag=134448175329440.
> CSeq: 1 INVITE.
> Call-ID: 134448175329440.fifouacctd.
> Content-Length: 226.
> User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
> Contact: <sip:caller at 50.57.54.156:5060 
> <http://sip:caller@50.57.54.156:5060>>.
> Content-Type: application/sdp.
> .
> v=0.
> o=click2dial 0 0 IN IP4 50.XX.XX.156.
> s=click2dial call.
> c=IN IP4 173.XX.XX.111.
> t=0 0.
> m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:97 ilbc/8000.
> a=rtpmap:98 speex/8000.
> #
> U 2012/08/08 22:09:13.766974 192.168.88.13:3072 
> <http://192.168.88.13:3072> -> 192.168.88.1:5060 
> <http://192.168.88.1:5060>
> SIP/2.0 404 Not found.
> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
> From: <sip:controller at ae.com 
> <mailto:sip%3Acontroller at ae.com>>;tag=134448175329440.
> To: <sip:9016726924 at irock.com <mailto:sip%3A9016726924 at irock.com>>.
> Call-ID: 134448175329440.fifouacctd.
> CSeq: 1 INVITE.
> User-Agent: snom821/8.7.3.10 <http://8.7.3.10>.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
> PRACK, MESSAGE, INFO, UPDATE.
> Allow-Events: talk, hold, refer, call-info.
> Supported: timer, replaces, from-change.
> Content-Length: 0.
>
> On Fri, Jul 27, 2012 at 3:24 PM, <duane.larson at gmail.com 
> <mailto:duane.larson at gmail.com>> wrote:
>
>     Very sure. Normal calls are working with clients behind the
>     OpenSIPS/SBC.
>
>
>
>
>     On , Bogdan-Andrei Iancu <bogdan at opensips.org
>     <mailto:bogdan at opensips.org>> wrote:
>     >
>     >
>     >
>     >
>     >
>     >
>     > Duane,some stupid question : are you sure your opensips is
>     > listening on the given IP:port ? have you check with netstat ?
>     > also have you checked with netstat also if there is traffic queued
>     > on the sockets ?
>     >
>     >
>     >
>     > Regards,
>     >
>     >
>     > Bogdan-Andrei Iancu
>     > OpenSIPS Founder and Developer
>     > http://www.opensips-solutions.com
>     >
>     >
>     > On 07/27/2012 12:48 AM, Duane Larson wrote:
>     > Oh yeah.  My first email has the SIPTrace from the
>     > OpenSIPS/SBC.  So I am logged into the OpenSIPS/SBC and did the
>     > NGREP.  So I see it SIP invite (99.XX.XX.161 is the IP of the
>     > OpenSIPS/SBC).   I would even let you log into the OpenSIPS/SBC
>     > and see it for yourself.  Makes no sense.
>     >
>     >
>     >
>     > U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 -> 99.XX.XX.161:5060
>     >
>     > INVITE sip:9016XXXXX at 192.168.88.13:3072;line=g2hfphrk SIP/2.0.
>     >
>     > Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.
>     >
>     > To: sip:9016XXXXX at irock.com <mailto:sip%3A9016XXXXX at irock.com>.
>     >
>     > From: sip:controller at ae.com
>     <mailto:sip%3Acontroller at ae.com>>;tag=134274001013257.
>     >
>     > CSeq: 1 INVITE.
>     >
>     > Call-ID: 134274001013257.fifouacctd.
>     >
>     > Content-Length: 155.
>     >
>     > User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
>     >
>     > Contact: sip:caller at 50.57.54.156:5060
>     <http://sip:caller@50.57.54.156:5060>>.
>     >
>     > Content-Type: application/sdp.
>     >
>     > .
>     >
>     > v=0.
>     >
>     > o=click-to-dial 0 0 IN IP4 0.0.0.0.
>     >
>     > s=session.
>     >
>     > c=IN IP4 0.0.0.0.
>     >
>     > b=CT:1000.
>     >
>     > t=0 0.
>     >
>     > m=audio 9 RTP/AVP 8 0.
>     >
>     > a=rtpmap:8 PCMA/8000.
>     >
>     > a=rtpmap:0 PCMU/8000.
>     >
>
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