[asterisk-bugs] [JIRA] (ASTERISK-21824) SIP INVITES from Telphin broke in 1.8.21

[Arcadiy Ivanov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21824?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=206881#comment-206881 ] 

Arcadiy Ivanov commented on ASTERISK-21824:
-------------------------------------------

This is a successful invite packet from Voicepulse:

10:37:18.735826 00:15:17:85:7b:a3 > 00:0c:f1:74:f7:ff, ethertype IPv4 (0x0800), length 1370: (tos 0x20, ttl  53, id 0, offset 0, flags [DF], proto: UDP (17), length: 1356) 64.61.93.190.5060 > 192.168.157.42.5060: [udp sum ok] SIP, length:
 1328

This is an invite packet from Telphin:

10:37:30.806243 00:15:17:85:7b:a3 > 00:0c:f1:74:f7:ff, ethertype IPv4 (0x0800), length 1514: (tos 0x20, ttl  46, id 35257, offset 0, flags [+], proto: UDP (17), length: 1500) 213.170.92.166.5068 > 192.168.157.42.5060: [bad udp cksum 45f6!] SIP, length: 1472

Things to note:

1) Telphin packet is 14 bytes larger than MTU.
2) Telphin has a bad checksum probably due to frag reassembly. I suspect the udp checksum offload/reassembly isn't working correctly on FreeBSD firewall for fragmented UDP packets.

At any rate, this is not an Asterisk issue.




                
> SIP INVITES from Telphin broke in 1.8.21
> ----------------------------------------
>
>                 Key: ASTERISK-21824
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21824
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.21.0
>         Environment: Linux pbx1.home.ivanovy.net 2.6.18-348.6.1.el5 #1 SMP Tue May 21 15:34:22 EDT 2013 i686 i686 i386 GNU/Linux
>            Reporter: Arcadiy Ivanov
>         Attachments: tcpdump.log
>
>
> Upgraded from 1.8.20 to 1.8.21. All SIP invites from Telphin no longer processed and are nowhere to be found in console with "sip set debug on", "core set verbose 999" and "core set debug 999". Other providers (Voicepulse and Gafachi) work fine.
> Out of all providers used in this system Telphin is the only one that has Record-Route in its invites. I highly suspect the issue may be with r379392 that changed the way Record-Route is parsed right after 1.8.20 was released.
> Attached is the tcpdump on the Asterisk box of the incoming SIP INVITES that are dropped silently.

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