a size of 2 bytes.
However we see that ALL RTP packets sent by the carrier side end point has
a length of 20 bytes.
Has anybody else seen this behavior from a carrier side endpoint ? Is
there an RFC or document that specifies
--
Thanks and Regards,
Vikram Ragukumar
Hello,
http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to
directly flow between endpoints and completely bypass Asterisk. I would
like to know if this configuration would work when,
a) both
or some other element along the SIP
message flow ? Does anybody know the difference in SIP message handling
between VoipSwitch and Asterisk or can anybody point me to an online
resource ?
--
Thanks and Regards,
Vikram Ragukumar
will be implementing encryption/decryption on the F.W
server.
Thanks and Regards,
Vikram.
Vikram Ragukumar wrote:
Hello,
- ---
|Sip Softphone|---|Internet||F.W|-|Asterisk
Hello,
- ---
|Sip Softphone|---|Internet||F.W|-|Asterisk|
- ---
IP addresses: a.b.c.dq.w.e.r
The SIP softphone(x-lite) is configured to
Steve Edwards wrote:
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
different boxes.
On Mon, 4 Jan 2010, Vikram Ragukumar wrote:
Would it be more efficient to use
Hello,
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
On Sun, 3 Jan 2010, Olle E. Johansson wrote:
No, Asterisk only supports one port.
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
, Vikram Ragukumar wrote:
Would it be more efficient to use libnetfilter_queue() to listen to
specific addresses / ports and forward to Asterisk?
Yes, but the number of SIP control messages are usually insignificant
compared to all the RTP packets.
If so, what would I be losing in not letting
