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?
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.
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.
Call-ID: 67108c50-1f0d-122e-17b1-001a647938bc.
CSeq: 382107 INVITE.
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.
.
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.