[Kamailio-Users] Possible timing/latency related problem ?

Vikram Ragukumar vragukumar at signalogic.com
Fri Apr 9 19:44:55 CEST 2010


Hello,

Alex, Juha thank you for your response. We have been doing more 
debugging here and have some more information regarding the problem

      Cell Phone     Kamailio       VoipSwitch
           |              |              |
           |INVITE        |              |
           |------------->|              |
           |100 Trying    |              |
           |<-------------|              |
           |              |INVITE        |
           |              |------------->|
           |              |100 trying    |
           |              |<-------------|
           |              |              |
           |INVITE        |              |
           |------------->|              |
           |100 Trying    |              |
           |<-------------|              |
           |              |183SessionProg|
           |              |<-------------|
           |              |180 Ringing   |
           |              |<-------------|
           |180 Ringing   |              |
           |<-------------|              |<-- 180 Ringing relayed before
           |183SessionProg|              |    183 Session Progress.
           |<-------------|              |    Proxy not rewriting c=
           |              |    200 OK    |    field in SDP before
           |    200 OK    |<-------------|    relaying to Cell phone
           |<-------------|              |
           |     ACK      |              |
           |------------->|              |
           |              |     ACK      |
           |              |------------->|
           |              |     BYE      |
           |              |<-------------|
           |    BYE       |              |
           |<-------------|              |

1) We setup logs for SIP message exchange at the proxy. The logs 
indicate that on "some occasions" 180 Ringing is relayed before 183 
Session Progress to the cellphone although these messages are received 
in the reverse order from the Sip server.

2) Connection parameter not rewritten in SDP of 180 Ringing. The c= 
connection parameter in the SDP of the 180 Ringing message that is 
relayed to the cellphone from the proxy has the ip address of the Sip 
server instead of the ip address of the proxy.

3) All other messages (183 Session Progress, 200 OK) that contain an SDP 
are rewritten by the proxy prior to relaying to the Cell phone.

Why is the SDP connection parameter not being rewritten for the 180 
Ringing message alone ?
How can i ensure the SIP messages are relayed in the order they are 
received ?

Thanks and Regards,
Vikram.
> Vikram Ragukumar writes:
> 
>  > In scenario 1 calls from the cellphone using a SIP softphone app go 
>  > through 100% of the time with both endpoints of the call being audible.
>  > However in scenario 2, calls from the cellphone go through with both 
>  > endpoints of the call being audible only sometimes, and during all other 
>  > attempts, call goes through with no voice from the cellphone.
> 
> i have noticed the same thing with some nokia phones.  if nokia receives
> 200 ok, it should start sending audio, but looks like sometimes it
> doesn't, since your rtpproxy is not receiving any.
> 
> have you checked if in those cases, nokia has sent ACK to kamailio or is
> it re-sending the invite?
> 
> -- juha





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