[SR-Users] One way communications in Kamailio 3.1.4

[Daniel-Constantin Mierla]

Hello,

On 9/14/11 6:13 PM, CaptWho wrote:
> Thanks Timo, I'll look into that, just have to figure out what you just said.
> :confused: I'm at the bottom of the learning curve and I got dumped into
> this after the guy that was dealing with it disappeared.
it will help also to watch the sip signaling traffic, using ngrep or 
wireshark.

For example with ngrep, on the same server with kamailio:

ngrep -d any -qt -W byline port 5060

will capture all the incoming and outgoing sip messages. You can see if 
there are some retransmissions, where the messages are sent, thus being 
able to detect if there is anything wrong in routing. If you do nat 
traversal, then also sdp and some headers must be updated in the proxy 
server.

Cheers,
Daniel

>
>
> Timo Klecker wrote:
>> Hello,
>>
>> the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is
>> not send E2E but using the Kamailio. In this case you would have to
>> redirect
>> it.
>>
>> In the content of INVITE and 200, 183 or ACK you should be able to see the
>> IPs for RTP data. Compare these with your phones.
>> O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX
>>
>> Be sure, the RTP ports are open on your firewall, too. You can see the
>> ports
>> in the content, too.
>> M=audio XXXXX RTP/AVP 8 101
>>
>> Greetings
>> Timo
>>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda