Quote:
Originally Posted by v164
However I've now encountered another SIP-incompatibility problem. My new Japanese VSP sends SIP INVITE packets where the final line of the SDP body ends abruptly without a terminating CRLF ( 0x0d 0x0a )....The AT-530 doesn't like that, and rejects the call with 488.
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"488 Not Acceptable Here"
In the meantime, a possible workaround would be to use VoXaLot's "Provider Registrations" feature to receive incoming calls from this VSP. Unlike the AT-530's current firmware, Voxalot's server doesn't seem to mind the SDP body with missing CRLF. Voxalot's server then forwards a properly-formatted SDP body to my phone in the SIP INVITE message (with a wider choice of codecs - although hopefully transcoding can be avoided).
However, I seem to have come across yet another SIP incompatibility problem.
VoXaLot successfully registers with the provider at ph2.so-net.ne.jp, and the provider appears to correctly use the "Contact: " URI - 660xxx@64.34.173.199:5061, however when a call comes in, the SIP INVITE message makes it all the way to my phone (which accepts the call, rings, and if I answer it, it even starts transmitting RTP packets), however the "100 Trying", "180 Ringing" and "200 OK" messages don't seem to make it back to my provider's SIP server.
The symptom is that, although my phone rings, the caller does not hear the ringing tone ("ringback").
The SIP invite packets from my provider arrive from 202.238.94.166, with a source port in the 30000 - 60000 range (eg. port 37422), however the "Via" header in the SIP INVITE wants the reply to be sent to port 5060. I thought that maybe VoXaLot was sending the "100 Trying", "180 Ringing" and "200 OK" messages back to port 37422 (which would be essential for "NAT Assistance"), and so the responses were being lost, however when I send a similar UDP packet to 64.34.173.199:5061 Voxalot correctly responds to port 5060.
I'm at a loss to explain it. Incoming calls work fine with X-Ten / Firefly SIP softphone, however the 100, 180, and 200 responses seem to go missing when the incoming call is processed through Voxalot. The same thing happens both when answering on my phone that's registered to Voxalot, and when my phone is un-registered, and Voxalot answers the call by voicemail.
Somewhere between 64.34.173.199:5061 and 202.238.94.166:5060 the 100, 180, and 200 SIP responses seem to go missing.