Hi Rashid,
I don't think RFC says that we can't send RTP after receiving INVITE and after
sending 100 Trying. Note as per RFC once we send offer in INVITE we must be
ready to receive RTP on any of the codec listed in the offer.
Please find extract from RFC 3264
" Once the offerer has sent the
Hello,
Quick question ...Is it possible that far end media gateway can send RTP right
after 100 TRYING. Means GW A sends an INVITE GW B respond with 100 TRYING and
start sending media (RTP) without sending 183 SESSION PROGRESS or 180 RINING
with SDP.
Let me know if this is correct also if i
If your proxy refrains from adding itself to the Record-Route header, it
will not remain in the chain. (You must take overt action to stay in.)
But note this is for *proxies*, not B2BUAs.
Paul
Mihaly Zachar wrote:
> Hi Gents,
>
>
> How is it possible to drop off a SIP chain after the
Hi Gents,
How is it possible to drop off a SIP chain after the call setup ?
I need to write a proxy which forwards the call from A to B, but after
the first transaction (INVITE, 18x, 200 OK, ACK) it should drop off the
SIP chain.
Any other messages should go between the 2 peers (A<->B).
Is it a
Hi,
Thank you for your responses. It seems that RFC 3261 has the following text
which confirms Paul's opinion that a race condition between BYE and
re-INVITE is not a glare...
"RFC 2543 was silent on whether a UA could initiate a new transaction to a
peer while another was in progress. That is n
