FreeSWITCH

No support for sip proxy forking

Details

  • Type: New Feature New Feature
  • Status: Closed Closed
  • Priority: Minor Minor
  • Resolution: Incomplete
  • Affects Version/s: None
  • Fix Version/s: None
  • Component/s: mod_sofia
  • Labels:
    None
  • Environment:
    freeswitch revision 9072
  • Platform:
    Linux x86/gcc
  • FreeSWITCH GIT Revision:
    unknown
  • Reproduced with GIT HEAD?:
    unknown

Description


The provider we are using is sending the following:

SDP from first 183 message from peer to freeswitch

v=0
o=Flowroute 3154 3154 IN IP4 66.234.182.42
s=sip call
t=0 0
m=audio 32264 RTP/AVP 0 101 13
c=IN IP4 66.234.182.42
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:13 CN/8000

SDP from second 183 message from peer to freeswitch

v=0
o=Flowroute 13797 3057 IN IP4 64.86.96.138
s=SIP Media Capabilities
t=0 0
m=audio 15262 RTP/AVP 0 101
c=IN IP4 64.86.96.138 <----------- different than first 183
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

SDP from 200 OK from peer to freeswitch

v=0
o=Flowroute 13797 3057 IN IP4 64.86.96.138
s=SIP Media Capabilities
t=0 0
m=audio 15262 RTP/AVP 0 101
c=IN IP4 64.86.96.138 <----------- different than first SDP, same as second SDP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

And freeswitch is sending the RTP data to IP it received in the first SDP, and ignoring the updated IP in the following SDP's.

The originate string has ignore_early_media=true.

I applied the patch from SFSIP-82 but it did not fix the problem. Have not tried to reproduce with
freeswitch trunk yet. (we are using revision 9072)

Activity

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Mike Jerris added a comment -
please attach the full sip trace of the call, not just the sdp
Show
Mike Jerris added a comment - please attach the full sip trace of the call, not just the sdp
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Traun Leyden added a comment -
Here is the sip trace exported as text.
Show
Traun Leyden added a comment - Here is the sip trace exported as text.
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Traun Leyden added a comment -
I talked to the providers sending us this SIP and got more information. What I understand is happening is that the call is getting "forked/branched" (I don't know this aspect of the SIP spec well so hopefully I have the terminology right)
 on their end as they are trying to connect the call to different upstream carriers simultaneously. So what is happening is that one carrier refuses the call, and then another one accepts.

Note that the to-tags are different on the 183's... hope that helps.
Show
Traun Leyden added a comment - I talked to the providers sending us this SIP and got more information. What I understand is happening is that the call is getting "forked/branched" (I don't know this aspect of the SIP spec well so hopefully I have the terminology right)  on their end as they are trying to connect the call to different upstream carriers simultaneously. So what is happening is that one carrier refuses the call, and then another one accepts. Note that the to-tags are different on the 183's... hope that helps.
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Mike Jerris added a comment -
we are waiting on the first part of the fix for this from the sofia-sip developer.
Show
Mike Jerris added a comment - we are waiting on the first part of the fix for this from the sofia-sip developer.
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John Riordan added a comment -
Might there be any news on this?
Show
John Riordan added a comment - Might there be any news on this?
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Marc Olivier Chouinard added a comment -
Mike, we got anything from the sofia-sip dev ?
Show
Marc Olivier Chouinard added a comment - Mike, we got anything from the sofia-sip dev ?
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Marc Olivier Chouinard added a comment -
Mike ?
Show
Marc Olivier Chouinard added a comment - Mike ?
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Jeff Lenk added a comment -
Please reopen if this issue is still valid or has not already been fixed.
Show
Jeff Lenk added a comment - Please reopen if this issue is still valid or has not already been fixed.
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Auto Admin added a comment -
Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours.
If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue.

Thanks,
  Jira Admin
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Auto Admin added a comment - Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours. If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue. Thanks,   Jira Admin
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Auto Admin added a comment -
Since there has been no change to the status of this issue, it is being closed for inactivity

Thanks,
  Jira Admin


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Auto Admin added a comment - Since there has been no change to the status of this issue, it is being closed for inactivity Thanks,   Jira Admin FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com

People

  • Assignee:
    Mike Jerris
    Reporter:
    Traun Leyden
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Dates

  • Created:
    Updated:
    Resolved: