Hello,
I try to configure Ser and polycom phones behind nat.
phone 1 ------- | ser | ----- internet phone 2 -------
Ser listen on private network and internet.
Polycom phones send public ip of Ser in SIP header. According to sip_call file (ethereal) ser know that phones are behind nat (lines 2,3,6,8) however when phone1 invite phone2 "404 Not found" message it sent by Ser .
If Ser Know that phones are behind nat How can I fix that problem in ser.cfg .
Thanks for your help.
Harry
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harry,
we tried polycom video phones a month or so ago and their sip implementation was worst i had seen so far. the phones didn't support digest authentication and we were also hit the problem you mentioned with domain in request-uri. i suggest to junk those phones before their sip implementation is closer to rfc3261.
-- juha
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc. g-)
Juha Heinanen wrote:
harry,
we tried polycom video phones a month or so ago and their sip implementation was worst i had seen so far. the phones didn't support digest authentication and we were also hit the problem you mentioned with domain in request-uri. i suggest to junk those phones before their sip implementation is closer to rfc3261.
-- juha
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Greger V. Teigre writes:
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc.
polycom phones were sent to us for testing by polycom dealer in finland. perhaps their software version was not current, but when we complained, no newer version was supplied. does yours support digest authentication of register and other requests and place callee's hostname (not ip address) in request uri?
-- juha
Never had any problem with digest auth, but yes, we had some problems with the IP address. However, here is the top of a current (after firmware upgrade) INVITE from a Polycom:
U 2005/05/27 10:50:42.733599 nat:2631 -> ser:5060 INVITE sip:number@ourdomain.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.2.2.54;branch=z9hG4bK7323c8ac29B7807. From: "Username" sip:userid@ourdomain.com;tag=9DFAF7D6-DB7B7E5B. To: sip:number@ourdomain.com;user=phone. CSeq: 2 INVITE. Call-ID: b07f3daa-4ce9e980-6bf54d45@10.2.2.54. Contact: sip:userid@10.2.2.54. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1.
As you can see, this is firmware 1.4.1.
Here's a REGISTER: U 2005/05/27 10:56:36.553574 nat:2631 -> ser:5060 REGISTER sip:ourdomain.com SIP/2.0. Via: SIP/2.0/UDP 10.2.2.54;branch=z9hG4bKf6e50949F4082504. From: "Username" sip:userid@ourdomain.com;tag=D548EE73-DCDBC98. To: sip:userid@ourdomain.com. CSeq: 845 REGISTER. Call-ID: 12d50ac7-c547e35d-a024f6c2@10.2.2.54. Contact: sip:userid@10.2.2.54;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER". User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1. Authorization: Digest username="userid", realm="ourdomain.com", nonce="4296dae8877bf0f3f27488cc28d24e60fe4f6ca3", uri="sip:ourdomain.com", response="a0bcf66d87eaf62a80beb1c975af3074", algorithm=MD5. Max-Forwards: 70. Expires: 3600. Content-Length: 0.
:-)
g-)
Juha Heinanen wrote:
Greger V. Teigre writes:
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc.
polycom phones were sent to us for testing by polycom dealer in finland. perhaps their software version was not current, but when we complained, no newer version was supplied. does yours support digest authentication of register and other requests and place callee's hostname (not ip address) in request uri?
-- juha
On 29-05-2005 11:08, Juha Heinanen wrote:
Greger V. Teigre writes:
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc.
polycom phones were sent to us for testing by polycom dealer in finland. perhaps their software version was not current, but when we complained, no newer version was supplied. does yours support digest authentication of register and other requests and place callee's hostname (not ip address) in request uri?
I spent quite some amount of time at sipits 2 years ago helping them to debug digest authentication and it was working after all with SER. I do not recall the phone model anymore -- if you have a picture of the phone I can tell you if it worked.
Jan.
Hello,
SER+nathelper+rtpproxy run on the same box that nat. Is it a problem ?
Harry
--- "Greger V. Teigre" greger@teigre.com a écrit :
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc. g-)
Juha Heinanen wrote:
harry,
we tried polycom video phones a month or so ago
and their sip
implementation was worst i had seen so far. the
phones didn't support
digest authentication and we were also hit the
problem you mentioned
with domain in request-uri. i suggest to junk
those phones before
their sip implementation is closer to rfc3261.
-- juha
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Running rtpproxy on a NATed box is a problem (you need a patch I posted a while back to announce the correct public IP). Running all daemons on the same server with a public IP is NOT a problem. g-) harry gaillac wrote:
Hello,
SER+nathelper+rtpproxy run on the same box that nat. Is it a problem ?
Harry
--- "Greger V. Teigre" greger@teigre.com a écrit :
That's strange. We are using some Polycom 300 and they work ok. No STUN support and sometimes a very erratic registration pattern, but except from that, they seem to perform ok. They interoperate with sipuras, grandstreams, sjphone, cisco gw, xlite, etc. g-)
Juha Heinanen wrote:
harry,
we tried polycom video phones a month or so ago
and their sip
implementation was worst i had seen so far. the
phones didn't support
digest authentication and we were also hit the
problem you mentioned
with domain in request-uri. i suggest to junk
those phones before
their sip implementation is closer to rfc3261.
-- juha
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com
-
Greger V. Teigre -
harry gaillac -
Jan Janak -
Juha Heinanen