[Freeswitch-users] Codecs (Again!)
Ahmed Sboor
ahmed at netelsat.net
Thu Apr 2 12:55:26 MSD 2015
Actually we had same issues with few soft switches and we resolved it by
using separate profiles at all , though absolute codec string also worked
perfect . Am Glad its done and thanks for sharing the cause of problem.
On Thu, Apr 2, 2015 at 12:05 PM, Andries Venter <apventer at gmail.com> wrote:
> I tried the settings for late negotiation without success (before I mailed
> the list!). I had an action application="answer" statement in my
> dialplan. This seems to override the whole negotiation process! After I
> removed that, the negotiation is done properly.
> Thank you Ahmed for the suggestion, even if it did not really solve the
> problem.
>
> On Wed, Apr 1, 2015 at 11:48 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Hi Andries ,
>> You should again look on
>> https://wiki.freeswitch.org/wiki/Codec_Negotiation
>>
>> And i think easiest way is using *absolute_codec_string *before briding
>> to "G729" only PABX.
>>
>> Ahmed
>>
>>
>> On Wed, Apr 1, 2015 at 7:51 PM, Andries Venter <apventer at gmail.com>
>> wrote:
>>
>>> I am using Freeswitch (1.4.8) as an SBC between our company PABX and our
>>> ISP's Trunk (also terminating on an SBC).
>>>
>>> My problem is that I need G729 in some cases and PCMA in others (the
>>> majority).
>>>
>>>
>>>
>>> The setup looks more or less like this:
>>>
>>> Our PABX (192.168.102.215)=>(192.168.102.102)Our SBC
>>> (10.17.159.10)=>(10.17.159.9)ISP SBC=>Rest of the world
>>>
>>>
>>>
>>> Since we use private addresses, Freeswitch must do RTP proxying. No
>>> transcoding must be done!!!
>>>
>>>
>>>
>>> I have set global_codec_prefs to "PCMA,G729". Then everything to
>>> destination supporting PCMA works. When trying to connect to a
>>> G729-only PABX, the call fails. Leg A is setup as follows
>>>
>>>
>>>
>>> 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 933
>>>
>>> INVITE sip:0873611337 at 192.168.102.102:5090;transport=udp SIP/2.0
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone>
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_iFGEOrlLLf
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>>>
>>> CSeq: 1235 INVITE
>>>
>>> Contact: <
>>> sip:anonymous at 192.168.102.215:5060;maddr=192.168.102.215>
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>>>
>>> Content-Type: application/sdp
>>>
>>> Content-Length: 288
>>>
>>> X-Siemens-Call-Type: ST-insecure
>>>
>>> Accept-Language: en;q=0.0
>>>
>>> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>>>
>>> Date: Wed, 01 Apr 2015 14:30:15 GMT
>>>
>>> Max-Forwards: 69
>>>
>>>
>>>
>>> v=0
>>>
>>> o=MxSIP 0 105140472 IN IP4 10.11.32.226
>>>
>>> s=SIP Call
>>>
>>> c=IN IP4 10.11.32.226
>>>
>>> t=0 0
>>>
>>> m=audio 5014 RTP/AVP 8 0 18 101
>>>
>>> a=rtpmap:8 PCMA/8000
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> a=rtpmap:18 G729/8000
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=silenceSupp:off - - - -
>>>
>>> a=fmtp:18 annexb=no
>>>
>>> a=fmtp:101 0-15
>>>
>>>
>>>
>>> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396
>>>
>>> SIP/2.0 100 Trying
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_iFGEOrlLLf
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone>
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>>>
>>> CSeq: 1235 INVITE
>>>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>>>
>>> Content-Length: 0
>>>
>>>
>>>
>>> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1201
>>>
>>> SIP/2.0 200 OK
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_iFGEOrlLLf
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone
>>> >;tag=BytX6X1t7HytF
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>>>
>>> CSeq: 1235 INVITE
>>>
>>> Contact: <sip:0873611337 at 192.168.102.102:5090;transport=udp>
>>>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>>>
>>> Accept: application/sdp
>>>
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>> Supported: timer, path, replaces
>>>
>>> Allow-Events: talk, hold, conference, presence,
>>> as-feature-event, dialog, line-seize, call-info, sla,
>>> include-session-description, presence.winfo, message-summary, refer
>>>
>>> Content-Type: application/sdp
>>>
>>> Content-Disposition: session
>>>
>>> Content-Length: 226
>>>
>>> Remote-Party-ID: "0873611337" <sip:0873611337 at 192.168.102.102
>>> >;party=calling;privacy=off;screen=no
>>>
>>>
>>>
>>> v=0
>>>
>>> o=FreeSWITCH 1427881297 1427881298 IN IP4 192.168.102.102
>>>
>>> s=FreeSWITCH
>>>
>>> c=IN IP4 192.168.102.102
>>>
>>> t=0 0
>>>
>>> m=audio 17318 RTP/AVP 8 101
>>>
>>> a=rtpmap:8 PCMA/8000
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=fmtp:101 0-16
>>>
>>> a=ptime:20
>>>
>>>
>>>
>>>
>>>
>>> When I change my global_codec_prefs to "G729,PCMA", G729 is offered to
>>> our PABX by Freeswitch, then to the ISP and then it works to destinations
>>> supporting G729. It fails to destinations not supporting G729. Now
>>> Leg A is setup as follows:
>>>
>>>
>>>
>>> 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 934
>>>
>>> INVITE sip:0873611337 at 192.168.102.102:5090;transport=udp SIP/2.0
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone>
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_8NZMvB3KNA
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>>>
>>> CSeq: 1235 INVITE
>>>
>>> Contact: <
>>> sip:anonymous at 192.168.102.215:5060;maddr=192.168.102.215>
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>>>
>>> Content-Type: application/sdp
>>>
>>> Content-Length: 289
>>>
>>> X-Siemens-Call-Type: ST-insecure
>>>
>>> Accept-Language: en;q=0.0
>>>
>>> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>>>
>>> Date: Wed, 01 Apr 2015 14:27:51 GMT
>>>
>>> Max-Forwards: 69
>>>
>>>
>>>
>>> v=0
>>>
>>> o=MxSIP 0 1224193414 IN IP4 10.11.32.226
>>>
>>> s=SIP Call
>>>
>>> c=IN IP4 10.11.32.226
>>>
>>> t=0 0
>>>
>>> m=audio 5014 RTP/AVP 8 0 18 101
>>>
>>> a=rtpmap:8 PCMA/8000
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> a=rtpmap:18 G729/8000
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=silenceSupp:off - - - -
>>>
>>> a=fmtp:18 annexb=no
>>>
>>> a=fmtp:101 0-15
>>>
>>>
>>>
>>> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396
>>>
>>> SIP/2.0 100 Trying
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_8NZMvB3KNA
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone>
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>>>
>>> CSeq: 1235 INVITE
>>>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>>>
>>> Content-Length: 0
>>>
>>>
>>>
>>> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1224
>>>
>>> SIP/2.0 200 OK
>>>
>>> Via: SIP/2.0/UDP 192.168.102.215:5060
>>> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>>>
>>> From: "JGD Winson" <sip:anonymous at anonymous.invalid
>>> >;tag=snl_8NZMvB3KNA
>>>
>>> To: <sip:0873611337 at 192.168.102.102;user=phone
>>> >;tag=g2eKXDUB6y4Hr
>>>
>>> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>>>
>>> CSeq: 1235 INVITE
>>>
>>> Contact: <sip:0873611337 at 192.168.102.102:5090;transport=udp>
>>>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>>>
>>> Accept: application/sdp
>>>
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>> Supported: timer, path, replaces
>>>
>>> Allow-Events: talk, hold, conference, presence,
>>> as-feature-event, dialog, line-seize, call-info, sla,
>>> include-session-description, presence.winfo, message-summary, refer
>>>
>>> Content-Type: application/sdp
>>>
>>> Content-Disposition: session
>>>
>>> Content-Length: 249
>>>
>>> Remote-Party-ID: "0873611337" <sip:0873611337 at 192.168.102.102
>>> >;party=calling;privacy=off;screen=no
>>>
>>>
>>>
>>> v=0
>>>
>>> o=FreeSWITCH 1427866745 1427866746 IN IP4 192.168.102.102
>>>
>>> s=FreeSWITCH
>>>
>>> c=IN IP4 192.168.102.102
>>>
>>> t=0 0
>>>
>>> m=audio 31726 RTP/AVP 18 101
>>>
>>> a=rtpmap:18 G729/8000
>>>
>>> a=fmtp:18 annexb=no
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=fmtp:101 0-16
>>>
>>> a=ptime:20
>>>
>>>
>>>
>>> I have tried various (hopefully all) combinations of
>>> inbound-codec-negotiation (generous, greedy and scrooge) and
>>> inbound-late-negotiation (true, false). Nothing does what I want.
>>>
>>>
>>>
>>> I believe Freeswitch needs to negotiate the codecs with the endpoint and
>>> choose one from my list. However, it seems to grab the first one in my
>>> list before checking the endpoint. Then it offers only that codec to
>>> the endpoint, which of course ignores / rejects it!
>>>
>>>
>>>
>>> What can I do?
>>>
>>> Regards
>>> Andries
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150402/90991518/attachment-0001.html
Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users
mailing list