[SIPForum-discussion] Call connected on different codec

[balajivenkat subramanian]

Guys ,

Here is the trace

Invite from ORG side

  Media Description, name and address (m): audio 37336 RTP/AVP 4 3 97 2 9
101
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:4 G723/8000
            Media Attribute (a): rtpmap:3 GSM/8000
            Media Attribute (a): rtpmap:97 iLBC/8000
            Media Attribute (a): fmtp:97 mode=20
            Media Attribute (a): rtpmap:2 G726-32/8000
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): ptime:60
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-11

183 response from Termination side

    Media Description, name and address (m): audio 17540 RTP/AVP 0 101
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15
            Media Attribute (a): ptime:20
            Media Attribute (a): silenceSupp:off - - - -

200 Ok from Termination side

      Media Description, name and address (m): audio 17540 RTP/AVP 0 101
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15
            Media Attribute (a): ptime:20
            Media Attribute (a): silenceSupp:off - - - -


On Thu, Dec 22, 2011 at 3:11 PM, Amit Salunkhe <amitsalunkhe21 at gmail.com>wrote:

> Hello Venkat
>
> If Customer INVITE contains G.711 ulaw codec in codec list then in this
> case call got connected and both way voice .
>
> Regrads
> Amit
>
> On Wed, Dec 21, 2011 at 5:11 AM, balajivenkat subramanian <
> balajivenkats at gmail.com> wrote:
>
>> Hello Guys ,
>>
>> Pls share your thoughts whether the below scenario is right behaviour .
>>
>> Customer  - > Softswitch - > Supplier .
>>
>> 1. Customer invite has codec G723 , ilbc etc .
>> 2.Supplier is responding back with only G711ulaw codec which was not
>> offered by Customer .
>> 3. Call got connected with voice on both sides .
>>
>> Thanks ,
>> Venkat
>>
>> _______________________________________________
>> This is the SIP Forum discussion mailing list
>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>> http://sipforum.org/mailman/listinfo/discussion
>> Post to the list at discussion at sipforum.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20111223/8dd6778f/attachment-0002.html>