Daniel,
I had posted this question earlier on asterisk-users, but didn't
receive any reply, so I'm posting on sr-users in the hope that someone
can provide guidance on how to debug this problem.
Our setup involves a Sip softphone registering with a fresh install of
Asterisk v1.6.0.5 through an
Klaus,
To me it seems that the SIP server also does some kind of NAT traversal:
it puts the Contact IP in to the RURI but it sends the request to the
IP:port from which the REGISTER was received (that's called NAT traversal).
So, either fix the SIP server (make sure it adds the port as in the
C
Klaus,
Thanks for your continued help.
How? Does it change the Contact header?
Yes, the proxy changes the Contact header.
So the SIP server is the registrar?
Correct.
The contact usually is the IP address of the client. So, if you the SIP
server routes based on the contact header, it sh
Klaus, Alex,
Thank you for your replies.
Does the SIP server perform NAT traversal?
Does the SIP proxy perform NAT traversal?
The SIP proxy performs NAT traversal.
How does the SIP server route the INVITE requests? Does it route based
on registered Contact or does it route them statically
Hello,
Alex, Klaus thank you for your replies.
Usually the softphones register to the proxy or SIP server. Thus, the
proxy or SIP server knows under which IP:port (The IP:port from which
the REGISTER was received) it can contact the sofpthone.
Yes, the softphones are registered to the SIP ser
Hello,
The diagram below shows the SIP message flow in our setup when Softphone
A places a call to Softphone B.
--- - -- - ---
|Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|Softphone B|
--- - --
Daniel,
you should set sip debug on for asterisk in order to get more verbose
output related to the 503 case. I couldn't spot something wrong for
register that got 503. Next ones have two contact headers, but that is
fine in SIP.
Thanks for your suggestion, I'm working on this and will get bac
Hello,
I had posted this question earlier on asterisk-users, but didn't receive
any reply, so I'm posting on sr-users in the hope that someone can
provide guidance on how to debug this problem.
Our setup involves a Sip softphone registering with a fresh install of
Asterisk v1.6.0.5 through an
Ovidiu,
Network jittter may have an effect on how much payload is available to be sent.
Take a look at this thread:
http://lists.rtpproxy.org/pipermail/users/2008-August/60.html
Check the value of POLL_LIMIT on the version of rtpproxy that you are using.
If you are able to run the same test
Ovidiu,
> If you have a codec that has silence suppression enabled, then you may
> get all kind of arbitrary lengths for packets (as silence suppression
> may kick in at any time).
> Disable silence suppression on both ends and retest. If you are still
> seeing variable length packets then there
Hello,
We are looking into bandwidth conservation by implementing RTP/UDP/IP
header compression.
Has anybody implemented ROHC or another header compression scheme in
combination with kamailio + rtpproxy ? Could you please point us to
online documentation or other useful resources ?
Thanks
Juha,
what was the conclusion regarding this? did the problem go away when
you called nat traversal functions both on 180 ringing and 183 session
progress?
Yes. The problem did go away once we started forcing rtpproxy on 180
Ringing also.
Thanks and Regards,
Vikram.
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