[asterisk-users] packet2packet bridging
Sameer Rathod
sameer at hostnsoft.com
Wed Jul 9 05:01:51 CDT 2014
Hi Ishfaq,
I am getting the the flow as attached Could you please read and check if
the rtp is passing directly as I am new and dont know much about this all
On Wed, Jul 9, 2014 at 3:24 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote:
> use tcpdump on the server to see if the RTP traffic is passing through it.
>
>
> On 9 July 2014 10:48, Sameer Rathod <sameer at hostnsoft.com> wrote:
>
>> Hi Mitul,
>>
>> I checked that the re-invite packet are sent what I want to check is
>> whether the audio packets is going through the server or not ?
>>
>>
>> On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani <mitul at enterux.in> wrote:
>>
>>> Put sip debug on to know if reinvite packets are sent.
>>> On 09-Jul-2014 1:17 PM, "Sameer Rathod" <sameer at hostnsoft.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Please clear me on this topic I am confused
>>>>
>>>> My log show "switching to native rtp".
>>>> Did this line means that the audio is not coming to the asterisk server
>>>> any more and asterisk only send the re- invite packet to both the clients ?
>>>>
>>>> Am I right or wrong ?
>>>>
>>>>
>>>> On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani <mitul at enterux.in>
>>>> wrote:
>>>>
>>>>> No way to avoid bw charges for any of the client if it is behind any
>>>>> sort of NAT.
>>>>> On 08-Jul-2014 8:52 PM, "Sameer Rathod" <sameer at hostnsoft.com> wrote:
>>>>>
>>>>>> Hi Eric,
>>>>>>
>>>>>>
>>>>>> I am behind nat
>>>>>>
>>>>>> Is there any solution for the same.
>>>>>>
>>>>>> My goal is to deduct the balance
>>>>>> for the call but free my asterisk server from audio packet load.
>>>>>>
>>>>>>
>>>>>> On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <EWieling at nyigc.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I think you will find that direct audio between two endpoints does
>>>>>>> not work when NAT is involved.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>>>>>>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Sameer
>>>>>>> Rathod
>>>>>>> *Sent:* Tuesday, July 08, 2014 11:18 AM
>>>>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>>> *Subject:* Re: [asterisk-users] packet2packet bridging
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Hi Joshua,
>>>>>>>
>>>>>>> I had disabled
>>>>>>>
>>>>>>> ice support and remover encryption= yes
>>>>>>>
>>>>>>> Then also it is showing the same native_rtp in log
>>>>>>>
>>>>>>> Could you help me in bypassing asterisk server for audio?
>>>>>>>
>>>>>>> please help me I am struggling with it form a long time.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod <sameer at hostnsoft.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> -- Channel SIP/1060-0000008e left 'native_rtp' basic-bridge
>>>>>>> <3c12ca41-e180-4fc1-80cf-1339b96da42b>
>>>>>>> -- Channel SIP/1061-0000008f left 'native_rtp' basic-bridge
>>>>>>> <3c12ca41-e180-4fc1-80cf-1339b96da42b>
>>>>>>> == Spawn extension (sameer, 1061, 1) exited non-zero on
>>>>>>> 'SIP/1060-0000008e'
>>>>>>>
>>>>>>> here are more generated when I cut the call
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod <sameer at hostnsoft.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> so In this case If I disable ice support
>>>>>>>
>>>>>>> ie commented the icesuppot=yes from all files
>>>>>>>
>>>>>>> then also I am getting this output
>>>>>>>
>>>>>>>
>>>>>>> -- Executing [1061 at sameer:1] Dial("SIP/1060-0000008e", "SIP/1061")
>>>>>>> in new stack
>>>>>>>
>>>>>>>
>>>>>>> == Using SIP RTP CoS mark 5
>>>>>>> -- Called SIP/1061
>>>>>>>
>>>>>>> -- SIP/1061-0000008f is ringing
>>>>>>> -- SIP/1061-0000008f answered SIP/1060-0000008e
>>>>>>> -- Channel SIP/1061-0000008f joined 'simple_bridge' basic-bridge
>>>>>>> <3c12ca41-e180-4fc1-80cf-1339b96da42b>
>>>>>>> -- Channel SIP/1060-0000008e joined 'simple_bridge' basic-bridge
>>>>>>> <3c12ca41-e180-4fc1-80cf-1339b96da42b>
>>>>>>> > Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from
>>>>>>> simple_bridge technology to native_rtp
>>>>>>> > 0x7f6800039020 -- Probation passed - setting RTP source
>>>>>>> address to 192.168.1.176:8000
>>>>>>> > 0x7f6780045810 -- Probation passed - setting RTP source
>>>>>>> address to 192.168.1.191:8000
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <jcolp at digium.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> Sameer Rathod wrote:
>>>>>>>
>>>>>>> yes I had configured
>>>>>>>
>>>>>>> icesupport=yes ;
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Asterisk does not support direct media establishment (with either
>>>>>>> chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Joshua Colp
>>>>>>> Digium, Inc. | Senior Software Developer
>>>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>>>>
>>>>>>> --
>>>>>>> _____________________________________________________________________
>>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>>>>>>> --
>>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>>> http://www.asterisk.org/hello
>>>>>>>
>>>>>>> asterisk-users mailing list
>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Sameer Rathod
>>>>>>>
>>>>>>> 8109413462
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Sameer Rathod
>>>>>>>
>>>>>>> 8109413462
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Sameer Rathod
>>>>>>>
>>>>>>> 8109413462
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> _____________________________________________________________________
>>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>>>>>>> --
>>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>>> http://www.asterisk.org/hello
>>>>>>>
>>>>>>> asterisk-users mailing list
>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards
>>>>>> Sameer Rathod
>>>>>> 8109413462
>>>>>>
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>> http://www.asterisk.org/hello
>>>>>>
>>>>>> asterisk-users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>> http://www.asterisk.org/hello
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards
>>>> Sameer Rathod
>>>> 8109413462
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>> http://www.asterisk.org/hello
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> Regards
>> Sameer Rathod
>> 8109413462
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: ish at pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
> 37 Ducie Street
> Manchester, M1 2JW
> COMPANY REG NO. 04920552
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Regards
Sameer Rathod
8109413462
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140709/09359d5f/attachment.html>
-------------- next part --------------
|Time | 192.168.1.191 | 68.233.237.171 |
| | | 111.118.250.237 | | 80.231.179.180 |
|15.667722| INVITE SDP (GSM speexRTPType-110 g711A g711U i...RTPType-98 t) | | |SIP From: <sip:101 at 111.118.250.237;transport=UDP To:<sip:918109413462 at 111.118.250.237;transport=UDP
| |(46865) ------------------> (5060) | | |
|15.673557| 401 Unauthorized | | |SIP Status
| |(46865) <------------------ (5060) | | |
|15.673956| ACK | | | |SIP Request
| |(46865) ------------------> (5060) | | |
|15.674035| INVITE SDP (GSM speexRTPType-110 g711A g711U i...RTPType-98 t) | | |SIP From: <sip:101 at 111.118.250.237;transport=UDP To:<sip:918109413462 at 111.118.250.237;transport=UDP
| |(46865) ------------------> (5060) | | |
|15.687637| 100 Trying| | | |SIP Status
| |(46865) <------------------ (5060) | | |
|16.410058| 183 Session Progress SDP (GSM g711U g711A tele...ne-eventRTPType-101) | | |SIP Status
| |(46865) <------------------ (5060) | | |
|17.250475| RTP (GSM) DTMF Seven 7 | | |RTP Num packets:1006 Duration:20.114s SSRC:0x5C24EB35
| |(8000) ------------------> (12340) | | |
|19.321498| RTP (GSM) | | | |RTP Num packets:857 Duration:17.179s SSRC:0x133699AC
| |(8000) <------------------ (12340) | | |
|30.002056| 180 Ringing | | |SIP Status
| |(46865) <------------------ (5060) | | |
|37.363429| RTP (GSM) | | | |RTP Num packets:9 Duration:0.072s SSRC:0x4D14F3B8
| |(8000) <------------------ (12340) | | |
|37.378048| 200 OK SDP (GSM g711U g711A telephone-eventRTP...e-101) | | |SIP Status
| |(46865) <------------------ (5060) | | |
|37.378682| ACK | | | |SIP Request
| |(46865) ------------------> (5060) | | |
|37.385289| RTP (GSM) | | | |RTP Num packets:1 Duration:0.000s SSRC:0x5C24EB35
| |(8000) ------------------> (12340) | | |
|37.387235| INVITE SDP (GSM g711U g711A telephone-eventRTP...e-101) | | |SIP Request
| |(46865) <------------------ (5060) | | |
|37.454019| 200 OK SDP (GSM speexRTPType-110 g711A g711U i...RTPType-98 t) | | |SIP Status
| |(46865) ------------------> (5060) | | |
|37.461000| ACK | | | |SIP Request
| |(46865) <------------------ (5060) | | |
|37.477267| RTP (GSM) | | | |RTP Num packets:117 Duration:2.328s SSRC:0x6FCA5C35
| |(8000) --------------------------------------> (18424) | |
|39.807179| INVITE SDP (GSM g711U g711A telephone-eventRTP...e-101) | | |SIP Request
| |(46865) <------------------ (5060) | | |
|39.873294| 200 OK SDP (GSM speexRTPType-110 g711A g711U i...RTPType-98 t) | | |SIP Status
| |(46865) ------------------> (5060) | | |
|39.882635| ACK | | | |SIP Request
| |(46865) <------------------ (5060) | | |
|39.895463| RTP (GSM) | | | |RTP Num packets:6226 Duration:124.511s SSRC:0x877D1683
| |(8000) ----------------------------------------------------------> (35798) |
|40.134393| RTP (g711U) | | |RTP Num packets:6174 Duration:124.677s SSRC:0x5BA486F0
| |(8000) <-------------------------------------- (18424) | |
|164.478037| BYE | | | |SIP Request
| |(46865) ------------------> (5060) | | |
|164.497298| 200 OK | | | |SIP Status
| |(46865) <------------------ (5060) | | |
More information about the asterisk-users
mailing list