[asterisk-bugs] [JIRA] (ASTERISK-23251) G729:60 packet size not being set correctly in SIP Invite

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23251?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=215413#comment-215413 ] 

Rusty Newton commented on ASTERISK-23251:
-----------------------------------------

You dialed direct to a SIP URI instead of using a SIP peer.

{noformat}
 -- Executing [999017 at default:1] Dial("SIP/10.10.19.28-00000014", "SIP/5555 at 10.10.4.58,,e") in new stack
{noformat}

{noformat}
hcc-dev1-avp*CLI> sip show peer peerHSRP
{noformat}

You showed the settings for peerHSRP, which unfortunately is not useful, since it wasn't involved in this call.


I noticed that in your description you said

bq. I've tried setting up specific SIP peers and also using the general codec settings, both experience the same issue.

So I went ahead and tried outbound calls, setting the RTP packetization in the global section like you had in your sip.conf and like you were doing in your debug log. That replicated the issue!

I then tested side-by-side, originating calls to peers set with "allow=g729:60" and non-peer SIP URI's like 6003 at ipaddress where the global section had "allow=g729:60". 

In my tests every call direct to a SIP URI would result in the codec from the global settings being used, but the RTP packetization setting not being applied.

For calls to peers in sip.conf, the RTP packetization was applied, with G729 and with other codecs.. so I wasn't able to reproduce it there.

Since you didn't show the issue with a call to a peer in your debug I wasn't able to gather anything further from that. If you can post debug (with the related sip.conf) for a call to a peer, where the issue occurs, then that would be interesting to see as well. If you do, be sure the debug log has the DEBUG type messages enabled and turned up to level 5. [see the wiki|https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information]

However, it looks like there is definitely a bug where RTP packetization is not applied to calls direct to a SIP URI with no peer involved, so I'm going to go ahead and open this up, then modify the summary and description.



                
> G729:60 packet size not being set correctly in SIP Invite
> ---------------------------------------------------------
>
>                 Key: ASTERISK-23251
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23251
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling, Codecs/General, Core/CodecInterface
>    Affects Versions: 1.8.11.1, 11.7.0
>         Environment: CentOS release 6.5 (Final) x64
> MySQL Database
> Dell R320 Server and Dell R210II Server
>            Reporter: Jarrod Sears
>            Assignee: Jarrod Sears
>            Severity: Critical
>         Attachments: asterisk-db.sql, asterisk-g72960-problem-nonrealtime-CLI.txt, asterisk-g72960-problem-nonrealtime-log.txt, g729_issue.log, nonrealtime-extensions.conf, nonrealtime-sip.conf, sip.conf
>
>
> Incoming calls to the server correctly negotiate to g729:60.
> The sip.conf is set to:
> disallow=all
> allow=g729:60
> The outbound leg of the server then sends a SIP invite out specifying g729:20.
> I've tried setting up specific SIP peers and also using the general codec settings, both experience the same issue.
> I have also tried using the SET(SIP_CODEC=G729:60) command, which does not work.
> I have tried using the a server with Digium's g729 codec. I've also tried on a server without a g729 codec. Both experience the same issue.
> I originally experienced the issue on the 1.8.11.1 (Asterisk 1.8.11-cert1) version that we are using generally for production. I recently installed a new copy of Asterisk 11.7.0 and have the same problem.

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