[SR-Users] Puzzling - INVITEs not making it to the fones

[Daniel-Constantin Mierla]

  Hello,

On 8/10/10 12:37 PM, Mahesh Paolini-Subramanya wrote:
> I'm seeing some *very* puzzling behavior from a bunch of our clients. 
>  Basically, the INVITE never makes it to the fone, it looks like the 
> upstream router/modem just swallows it.  The weird thing is, tweaking 
> the username slightly resolves the issue, for a while at least.  And 
> then, it stops again.   Anyone seen anything like this before?  Any 
> ideas?  Am I missing something ridiculously obvious?
> Fwiw, I've seen this behavior in setups with *no* SIP ALG, and *no* 
> firewall.  That said, it really *does* look like a firewalling issue, 
> doesnt it?

if the request is sent by kamailio but not reaching the phone, then is 
something in the middle inspecting SIP and discarding (big brother?). Is 
it a particular type of phone, or any phone? At first sight, the request 
seems just fine. I cannot think of something else than alg/firewall in 
the middle, unless the phone itself discards it.

Cheers,
Daniel

>
> Here is a typical INVITE that is sent out from Kamailio.  The username 
> is /mahesh.test204/.  The "fix" is to simply change the name to 
> "mahesh.test205", or "mahesh.test20", or "ahesh.test204", etc.
> Is there something obviously wrong with the INVITE perhaps?  Bad format?
>
>
>
> INVITE 
> sip:mahesh.test204 at 173.101.106.53:5060;rinstance=e2f54e94d29118d7;transport=UDP 
> SIP/2.0.
> Record-Route: <sip:74.217.82.147;lr;ftag=X8BN4N8vtXKej;nat=yes>.
> Via: SIP/2.0/UDP 74.217.82.147;branch=z9hG4bK8e0c.0ded64d5.0.
> Via: SIP/2.0/UDP 
> 74.217.82.232;received=74.217.82.232;rport=5060;branch=z9hG4bK3DtFg45Qetv9F.
> Max-Forwards: 69.
> From: "7732206484" <sip:2065771035 at 74.217.82.232>;tag=X8BN4N8vtXKej.
> To: <sip:mahesh.test204 at a5prodsbc.aptela.com>.
> Call-ID: 67108c50-1f0d-122e-17b1-001a647938bc.
> CSeq: 382107 INVITE.
> Contact: <sip:mod_sofia at 74.217.82.232:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla, 
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 246.
> Subject: Local.
> Remote-Party-ID: "7732206484" 
> <sip:2065771035 at 74.217.82.147>;party=calling;id-type=subscriber;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1281422601 1281422602 IN IP4 74.217.82.232.
> s=FreeSWITCH.
> c=IN IP4 74.217.82.232.
> t=0 0.
> m=audio 13742 RTP/AVP 0 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
>
>
>
> Mahesh Paolini-Subramanya | CTO | mahesh at aptela.com 
> <mailto:mahesh at aptela.com> | 703.386.1500 Ext. 9100
> 2250 Corporate Park Drive | Suite 150 | Herndon, VA | www.aptela.com 
> <http://www.aptela.com/>
> Check out our Blog <http://www.aptela.com/blog> | Follow us on Twitter 
> <http://twitter.com/Aptela> | Refer a Friend 
> <http://www.aptela.com/referral>
>
>
>
> _______________________________________________
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-- 
Daniel-Constantin Mierla
http://www.asipto.com/

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