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New Member

SIP Call Flow - Invite from ISP after RTP established

Hi,

I have the following issue: when I try to dial a PSTN number, from the customer, using the G729 the call is not established and I'm getting a fast busy. The ISP is telling us that we should be sending a 488 after their Invite w/SDP.

What I find strange is that after the RTP seems to be established there is a new INVITE SDP from the ISP requesting the call to be done in g729 annexb.

Here's the call flow (1.1.1.1 is the Customer and 99.99.99.99 the IMS Provider):

 

|Time     | 1.1.1.1                         |
|         |                   | 99.99.99.99      |                  
|3,174    |         INVITE SDP (g729 telephone-eventRTPType-101 CN...d))          |SIP From: "Ricardo Godinho" <sip:211135916@1.1.1.1 To:<sip:217231800@99.99.99.99
|         |(60083)  ------------------>  (5060)   |
|3,176    |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|3,765    |         180 Ringing                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|3,831    |         200 OK SDP (g729 CN(old) telephone-eventRTPTyp...01)          |SIP Status
|         |(5060)   <------------------  (5060)   |
|3,832    |         ACK       |                   |SIP Request
|         |(60083)  ------------------>  (5060)   |
|3,935    |         RTP (g729)                    |RTP Num packets:4  Duration:0.059s SSRC:0xBB2BD6C7
|         |(19206)  ------------------>  (15872)  |
|4,012    |         INVITE SDP (g729 telephone-eventRTPType-101 CN...d))          |SIP Request
|         |(5060)   <------------------  (5060)   |
|4,013    |         100 Trying|                   |SIP Status
|         |(60083)  ------------------>  (5060)   |
|4,021    |         BYE       |                   |SIP Request
|         |(60083)  ------------------>  (5060)   |
|4,033    |         200 Race Condition            |SIP Status
|         |(5060)   <------------------  (5060)   |
|12,014   |         CANCEL    |                   |SIP Request
|         |(5060)   <------------------  (5060)   |
|12,014   |         481 Call Leg/Transaction Does Not Exist          |SIP Status
|         |(60083)  ------------------>  (5060)   |
 

Any help?

 

Thanks.

  • IP Telephony
5 REPLIES
VIP Super Bronze

You need to go back to them.

You need to go back to them. "488 means media unacceptable", I am not sure  they expect you to send that after a RE-INVITE. They shouldn't send a 200 OK to your INVITE if they cant do g729. You may also want to offer them G729 annexb if thats the only flavor of g729 they support

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Thanks for the reply. They do

Thanks for the reply.

 

They do support g729, however it seems that after the 200 Ok, they are sending another SIP Invite to change it to G729 annexb. Do you mean adding the g729br8 on the voice class codec?

 

VIP Super Bronze

Yes, I suggest you try that,

Yes, I suggest you try that, but you may want to find out why they are sending a RE-INVITE after 200 ok..What is it they dont like in your offer?

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

I'm asking that to the ISP.

I'm asking that to the ISP. Let's see...if they have an explanation for that.

New Member

I have removed the codecs

I have removed the codecs that are not being used from the offer and the problem was resolved. So the annex b was removed from the voice class codec.

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