Hello,

When I make a call from a Siemens handset to SJPhone (softphone), I notice
something strange in the "To:" field in the INVITE request.

This field looks like this:

To: <sip:192.168.0.247:5060;user=phone>

Is this really a valid notation?? I am writing a parser that reads these
headers but I am unsure how to interpret this...

I would expect the To field to look like this:
To: <sip:192.168.0.247:5060>;user=phone


Cheers,
Alex


15:45:53 DEBUG
2010-05-19 13:45:53.656 UDP 192.168.0.2:5070->LOCAL
INVITE sip:192.168.0.247:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5070
;branch=z9hG4bKe689123fe2f49092f291e02e3b31823a;rport
From:  <sip:192.168.0.2:5070>;tag=3496067849
To: <sip:192.168.0.247:5060;user=phone>
Call-ID: 2858486...@192_168_0_2
CSeq: 2 INVITE
Contact: <sip:[email protected]:5070>
Max-Forwards: 70
User-Agent: C470IP022230000000
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 351

v=0
o=31237112291 8194 138 IN IP4 192.168.0.2
s=Mapping
c=IN IP4 192.168.0.2
t=0 0
m=audio 8194 RTP/AVP 0 8 96 97 2 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
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