how can proxies edit c line in sdp? which rules can be applied to c line by
proxies?
I have a sip client behind nat which insert its own private IP at session
level (c line under m line)
and NAT adds its own public IP into c line at media level before forwarding
200 OK to proxy.
Here, proxy remove
On Mon, 2009-01-05 at 22:38 +, Andrew Wood wrote:
> Im trying to implement a simple forking proxy server.
> In the example below the calling phone (200) is on 192.168.254.1
> The called phone (201) is on 192.168.254.2
> The proxy is at 192.168.254.254
>
> Following is the Invite received by th
> Is one correct and one not or does it not
> matter as long as they both match?
RFC 3261 defines the user-param within a sip-uri. It also defines how
to build responses; the response shown is not built correctly.
For completeness... it would also be valid to place user=phone as a
to-param with
user= is a SIP URI parameter and hence must be included in angle brackets.
Therefore the one in INVITE is correct.Otherwise in 180 there is nothing
differentiating tag= and user= parms.
2009/1/6 Andrew Wood
> Thanks Brett. Is one correct and one not or does it not matter as long
> as they both m
Thanks Brett. Is one correct and one not or does it not matter as long
as they both match?
Regards
Andrew
Sent from iPhone
On 5 Jan 2009, at 22:46, "Brett Tate" wrote:
> There is a user=phone issue within the 180's To header. It is inside
> the brackets for INVITE; it is outside for 180.
>
I agree with Maxim here. You should try to classify traffic to various
destinations and configure application timers (possibly in a table) to these
destinations.A logging mechanism can help you fine tune these values and
find new destinations that have high RTT and hence are timing out.
2009/1/3 M
Hi,
Thanks for your reply, I understand that for outgoing call we can do this but
ports will be open for limited period of time but if assume I want to receive
call then how to do that?
I guess for that I require STUN client in my UA and STUN server in my
Application server. My question is whe
Hi,
I am doing Load testing with SIPP , But now setup got changed and I want
to test SIP behind NAT. Can any one please tell me whether SIPP works with NAT
? If not can any one point me to the tool. Where I can generate Load for SIP
behind NAT.
Regards
Chandra sekhar
_
