[Asterisk-Dev] About RTP session starting in Asrerisk

[phu at bevertec.com]

Hi, group:

I found a problem with Asterisk as proxy to test sip-comminicator by
monitoring underlying sip interaction.

Once  sip-communicator(56) initializes a call to destination(205) through
Astersik(22), Asterisk responses with OK including SDP, but "Connection
Information" in SDP is Asterisk proxy's IP address(22) RATHER THAN
destination UAS's IP(205),

After UAC send ACK to Asterisk, Asterisk sends back one more Invite with
SDP to UAC. It is in THIS SDP that includes UAS's IP address as following:

SIP-COMMUNICATOR(56) initialize a call to x-lite(205) through Asterisk(22):

  num      56(sip-communiator)      22(Asterisk)           205(X-lite)


                       Invite/SDP
   1         |-------------------------->|
             |                           |
   2         |          407 Auth         |                        |
             |<--------------------------|                        |
             |                           |                        |
             |        ACK                |                        |
   3         |-------------------------->|                        |
             |                           |                        |
             |     Invite(SDP/56)        |                        |
   4         |-------------------------->|     Invite(SDP/22)     |
             |                           |----------------------->|
             |         Trying            |                        |
   5         |<--------------------------|                        |
             |                           |                        |
             |       Ringing             |                        |
   6         |<--------------------------|                        |
             |                           |          ACK           |
             |      OK(SDP/22)           |<-----------------------|
   7         |<--------------------------|                        |
             |                           |        Invite(SDP/56)  |
             |        ACK                |----------------------->|
   8         |-------------------------->|                        |
             |                           |                        |
             |       Invite(SDP/205)     |                        |
   9         |<--------------------------|                        |
             |                           |         ACK            |
             |         Trying            |<-----------------------|
  10         |-------------------------->|                        |
             |                           |                        |
             |                           |                        |
             |     [ OK(SDP/56 ) ]       |                        |
  11         |-------------------------->|                        |
             |                           |                        |
             |       [ ACK ]             |                        |
  12         |<--------------------------|                        |
             |                                                    |
             |                                                    |
             |                          RTP                       |
             |--------------------------------------------------->|
             |                          RTP                       |
             |<---------------------------------------------------|





>From above, I think that Asterisk doesn't abide by RFC3261, "Connection
Infomation" in SDP with OK response[7] should be UAS IP address, instead
of Asterisk proxy's IP.
Asterisk should NOT send back one more Invite request to UAC. RTP session
should start right after UAC send ACK to Astersik

Thereofre, some UAs abiding to RFC3261 such sip-communicator are
impossible to work with Astersik.

How do you think about it?


Thanks for any response from you!



Paul