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Summary:ASTERISK-02414: RTP is not immediately transmitted after 183 session progress is sent
Reporter:tschroer (tschroer)Labels:
Date Opened:2004-09-15 13:30:56Date Closed:2011-06-07 14:10:27
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Situation: A call comes into the * from Carrier A and sent to a user agent that is sent to respond back with a 302

message. * sends Carrier A a 180 Ringing response.  User Agent sends back a 302 Moved Temporarily message.  *

proceeds to establish another call with the new number in the contact field of the 302 response and sends the INVITE

to Carrier B.  Carrier B sends a 183 Session Progress Message.  * Sends a Session Progress to Carrier A.

Here is where the issue is.  While * sends out a 183 message to Carrier A once it receives the 183 from Carrier B,

it does not start sending RTP to Carrier A until an OK is received first from Carrier B.  It would seem that *

should be sending RTP to Carrier A immediately up sending a 183 Message to Carrier A with SDP of media.  With this

particular soft switch vendor, if it does not receive RTP after receiving a 183 it cancels the call.  

Solution: if * receives a 183 Session Progress from Carrier B AND corresponding RTP (which it does), * should start

sending that RTP to Carrier A after it sends the 183 message to it.

Carrier A:       10.10.10.1
Carrier B:       204.154.10.10
Asterisk:        172.16.100.10
Registrar Proxy: 192.168.100.1
(NOTE: Substituted IP addresses)


10.10.10.1 172.16.100.10 SIP/SDP Request: INVITE sip:9725551212@172.16.100.10:5060, with SDP

172.16.100.10 10.10.10.1 SIP Status: 100 Trying

172.16.100.10 192.168.100.1 SIP/SDP Request: INVITE sip:9725551212@proxy.golinx.net, with SDP

172.16.100.10 10.10.10.1 SIP Status: 180 Ringing

192.168.100.1 172.16.100.10 SIP Status: 100 Trying

192.168.100.1 172.16.100.10 SIP Status: 302 Moved Temporarily

172.16.100.10 192.168.100.1 SIP Request: ACK sip:8165551212@proxy.golinx.net

172.16.100.10 204.154.10.10 SIP/SDP Request: INVITE sip:18165551212@204.154.10.10, with SDP

204.154.10.10 172.16.100.10 SIP Status: 100 Trying

204.154.10.10 172.16.100.10 SIP/SDP Status: 183 Session Progress, with SDP

172.16.100.10 10.10.10.1 SIP/SDP Status: 183 Session Progress, with SDP

10.10.10.1      172.16.100.10   SIP     Request: CANCEL sip:9725551212@172.16.100.10:5060

172.16.100.10   10.10.10.1      SIP      Status: 200 OK

The CANCEL message is sent depending upon when * receives the OK from Carrier B.  If it is received quickly enough and sent to Carrier A, then there is no CANCEL issued.  Carrier A (Level 3) is using this restriction as a means for detection of rogue RTP.
Comments:By: Mark Spencer (markster) 2004-09-15 13:41:34

I believe this is a configuration problem, most likely caused by your using the "r" option in the Dial, can you please post the dial entry you used?
By: tschroer (tschroer) 2004-09-15 19:10:22

Sorry for the delay in getting back but wanted to retest.  Went to extensions.conf file and took out the "r" option and rtp started immediately after the 183 message.  Also, went to the voip-info.org and better scrutinized the dial() command description page and saw the warning in the the options parameter description!!!!!  Thanks for pointing this out.

Case Closed

Tom