Hi Tony,

The cause of the 488 is now known.   It is the fact that when SRTP is 
enabled, Polycom offer two m-lines.  One for the enrypted stream, and 
the other the normal none-encrypted stream.  Both are offered using the 
same port.  The normal answer from a compliant provider is to offer a 
port with "0" value for the encrypted stream while sending an actual 
port for the none encrypted stream (given that it doenst know how to 
support the SRTP attributes).


However, there is yet another bug discovered in sipXbridge.  It only 
rewrites or process the first m-line and leaves the the second m-line 
untouched.  The other bug is that it choose to process the first m-line 
(the encrypted one) and rewrote its port.  however it omitted the SRTP 
attributes rendering that stream corrupted even if negotiation succeeds. 
We will be opening a new task for this.

*** Take note that the offer from sipXbridge is still valid and must not 
generate a 488.  Only incompatible streams.  This reveals that your 
provider blindly generates a 488 if its sees multiple m-lines for audio 
instead of choosing between them.

disabling SRTP as the default should be the conclusion of 9699.  We will 
reference the new ticket in it.

joegen

On 06/22/2011 08:05 PM, Tony Graziano wrote:
> I saw srtp in the siptrace, but not in the pcap (generated from
> firewall), so it never made it to the itsp. I also removed it from the
> phone and then had 2 way audio.
>
> If you look at both siptraces, they are markedly different.
>
> In looking at the polcyom settings in sipxconfig it defaults to:
>
> Enable SRTP            (Default: checked)
> If checked, the phone accepts SRTP offers.
> Offer SRTP                 (Default: checked)
> If checked, the phone offers SRTPs.
>
> I disabled that and then audio worked and the other POPS started to work.
>
> I would encourage you to look at the two siptraces, as the
> transactions are very much  different, where they should be nearly
> identical except for the POP and transaction ID. I think the default
> srtp should be off, and a wiki entry needs to explain how to properly
> test it before using it, because it seems to have an unintended
> consequence out of the box.
>
> On Wed, Jun 22, 2011 at 7:53 AM, George Niculae<[email protected]>  wrote:
>> On Wed, Jun 22, 2011 at 2:47 PM, Tony Graziano
>> <[email protected]>  wrote:
>>> I removed that from the bridge, no luck. Works with G722 in some pops,
>>> not all. I also removed that from the phone. IF you look at the
>>> siptrace examples, there is no "moved temporarily" on the failed call.
>>>
>>> Also, with polycom remote users, I now have one way audio on
>>> successful calls (remote users cannot be heard but can hear) . I have
>>> 2 way audio with xlite for remote users (same network, etc.).
>>>
>>> So I feel whatever "other" changes got put in (like maybe srtp stuff)
>>> for polycom may have had an unintended consequence.
>>>
>> Before reverting, just to be sure srtp is or is not the culprit, can
>> you go in sipxconfig to polycom>  security and uncheck srtp option,
>> then try one more time?
>>
>> Thanks
>>
>>> It would be really helpful to be able to rollback to 4.4 patch #1 or #2...
>>>
>>> Is there a way to get back to patch level 1 or 2 until this gets
>>> sorted out? Otherwise I can't see the likelihood of any successful 4.4
>>> install right now.
>>>
>>> On Wed, Jun 22, 2011 at 6:29 AM, Joegen Baclor<[email protected]>  wrote:
>>>> Hi Tony,
>>>>
>>>> Definitely coincidental.  The call ended due to a codec issue.  Probably
>>>> G.722?
>>>>
>>>> Joegen
>>>>
>>>>
>>>> On 06/22/2011 06:07 PM, Tony Graziano wrote:
>>>>> I posted pcaps and siptraces of a failed/successful transaction in
>>>>> outbound calls. Hopefully this will shed some light on whether this is
>>>>> related to the patch or not.
>>>>>
>>>>> On Wed, Jun 22, 2011 at 5:49 AM, Tony Graziano
>>>>> <[email protected]>    wrote:
>>>>>> strangely I noticed I cannot place outbound calls with one provider
>>>>>> starting yesterday. This coincided with when i applied the patch (#6)
>>>>>> from the build repo onto the production system. At the same time I can
>>>>>> place calls with this provider through a different POP for them. I
>>>>>> have sent them pcaps of a failed and working call to see if it is
>>>>>> their problem. Depending on their response, I may add to the JIRA and
>>>>>> not close it.
>>>>>>
>>>>>> On Wed, Jun 22, 2011 at 4:28 AM, Tony Graziano
>>>>>> <[email protected]>    wrote:
>>>>>>> I'd like to hear back from Dragan and Josh who were experiencing the
>>>>>>> problems or had a system that was, but if we don;t hear back from them
>>>>>>> by the end of the day I'll close it.
>>>>>>>
>>>>>>> On Wed, Jun 22, 2011 at 12:02 AM, Joegen Baclor<[email protected]>
>>>>>>>   wrote:
>>>>>>>> Shall we mark the task as resolved so you can close it?
>>>>>>>>
>>>>>>>> On 06/22/2011 02:37 AM, Tony Graziano wrote:
>>>>>>>>
>>>>>>>> so far that seems to have fixed it.
>>>>>>>>
>>>>>>>> On Jun 21, 2011 10:03 AM, "Douglas Hubler"<[email protected]>    wrote:
>>>>>>>>> On Tue, Jun 21, 2011 at 8:49 AM, Douglas Hubler<[email protected]>
>>>>>>>>>   wrote:
>>>>>>>>>> On Tue, Jun 21, 2011 at 8:19 AM, George Niculae<[email protected]>
>>>>>>>>>>   wrote:
>>>>>>>>>>> There is a fix committed by Joegen and Douglas triggered RPMs build,
>>>>>>>>>>> we'd really appreciate if you could update and give it a try later
>>>>>>>>>>> today when available
>>>>>>>>>> I will post link to build when available
>>>>>>>>> http://ec2-50-17-110-224.compute-1.amazonaws.com:40100/
>>>>>>>>>
>>>>>>>>> I'll only keep it up for few hours or until someone tells me it's
>>>>>>>>> good, then i'll push to download.sipfoundry.org. if for some reason
>>>>>>>>> there is still a problem with this build, don't worry, you will still
>>>>>>>>> be able to update from this build to the final build.
>>>>>>>>> _______________________________________________
>>>>>>>>> sipx-users mailing list
>>>>>>>>> [email protected]
>>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>>>>>> _______________________________________________
>>>>>>>> sipx-users mailing list
>>>>>>>> [email protected]
>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> ======================
>>>>>>> Tony Graziano, Manager
>>>>>>> Telephone: 434.984.8430
>>>>>>> sip: [email protected]
>>>>>>> Fax: 434.326.5325
>>>>>>>
>>>>>>> Email: [email protected]
>>>>>>>
>>>>>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>>>>>> Telephone: 434.984.8426
>>>>>>> sip: [email protected]
>>>>>>>
>>>>>>> Helpdesk Contract Customers:
>>>>>>> http://support.myitdepartment.net
>>>>>>> Blog:
>>>>>>> http://blog.myitdepartment.net
>>>>>>>
>>>>>>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>>>>>>
>>>>>>
>>>>>> --
>>>>>> ======================
>>>>>> Tony Graziano, Manager
>>>>>> Telephone: 434.984.8430
>>>>>> sip: [email protected]
>>>>>> Fax: 434.326.5325
>>>>>>
>>>>>> Email: [email protected]
>>>>>>
>>>>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>>>>> Telephone: 434.984.8426
>>>>>> sip: [email protected]
>>>>>>
>>>>>> Helpdesk Contract Customers:
>>>>>> http://support.myitdepartment.net
>>>>>> Blog:
>>>>>> http://blog.myitdepartment.net
>>>>>>
>>>>>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>>>>>
>>>>>
>>>>
>>>
>>>
>>> --
>>> ======================
>>> Tony Graziano, Manager
>>> Telephone: 434.984.8430
>>> sip: [email protected]
>>> Fax: 434.326.5325
>>>
>>> Email: [email protected]
>>>
>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>> Telephone: 434.984.8426
>>> sip: [email protected]
>>>
>>> Helpdesk Contract Customers:
>>> http://support.myitdepartment.net
>>> Blog:
>>> http://blog.myitdepartment.net
>>>
>>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>> _______________________________________________
>>> sipx-users mailing list
>>> [email protected]
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
>> _______________________________________________
>> sipx-users mailing list
>> [email protected]
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>

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