[Freeswitch-users] SDP with double m=audio lines and with the same port

Ján Füri furi at vmtele.com
Tue May 31 21:04:49 MSD 2016


I asked anybody whether there is a way how to achieve my goal. Somebody 
that is more skilled than I am.
I was asking if there are "trivially easy ways to accomplish my goal". 
Freeswitch is an awesome software, I like it very much, but replacing 
kamailio with freeswitch is not a solution for me (in our environment).


On 31.05.2016 18:28, Michael Jerris wrote:
> Its all software, anything is possible with enough code modification. 
>  We don't have a setting to do those things right now, as it doesn't 
> make any sense when there are trivially easy ways to accomplish the 
> same goal without writing any more code.
>
>
>> On May 31, 2016, at 12:07 PM, Ján Füri <furi at vmtele.com 
>> <mailto:furi at vmtele.com>> wrote:
>>
>> Thank you Michael,
>> I understand, so isn't it possible to have SDP with m=audio lines 
>> SAVPF, SAVP and AVP all together ?
>> Because I set media_webrtc=true, AVP and SAVP lines are replaced with 
>> SAVPF.
>> If I had all three m=audio proto lines I was able to manage it with 
>> Kamailio.
>>
>> And back to my first question, is that possible to set different 
>> ports  for m=audio lines in SDP ?
>>
>> Jan
>>
>> On 31.05.2016 17:16, Michael Jerris wrote:
>>> If you don't use some other engine to handle webrtc, then if you are 
>>> calling something that is registered to you over websockets it will 
>>> automatically enable media_webrtc for you.  Otherwise you'll need 
>>> some external way of knowing if the endpoint is webrtc or not so you 
>>> can apply the settings properly.
>>>
>>>
>>>> On May 30, 2016, at 9:14 AM, Ján Füri <furi at vmtele.com> wrote:
>>>>
>>>> Hello,
>>>>
>>>> My freeswitch's INVITE sdp offers two m=audio lines. (RTP/SAVP and 
>>>> RTP/AVP).
>>>> That's OK, because I want it so ;)
>>>>
>>>> But both use the same audio port number.
>>>> Is that possible to change this behavior ? To use different audio 
>>>> port numbers for each m=audio line ...
>>>> Please check my example below. In SDP from my freeswitch, both 
>>>> m-lines have *m=audio **22284*.
>>>>
>>>> The reason, why I want to change this is WebRTC and RTPengine. 
>>>> RTPengine changes this INVITEs to RTP/SAVPF.
>>>> And the changed RTP/SAVPF is then not acceptable for chrome 
>>>> browsers. Firefox works well (so far).
>>>> I already reported this to the RTPengine but they say the problem 
>>>> is with INVITE from freeswitch because the m-audio lines use the 
>>>> same port.
>>>>
>>>> My example :
>>>> incoming calls -> Media Server (freeswitch) -> Kamailio (sip and 
>>>> websocket proxy) -> WebRTP and SIP clients
>>>>
>>>> I have found a feature *media_webrtc=true* so I could avoid using 
>>>> rtpengine for webrtc clients (that would be really awesome), but 
>>>> then classic sip clients are offered only with RTP/SAVPF and most 
>>>> sip phones do not know RTP/SAVPF so the calls are rejected :(
>>>>
>>>>
>>>> INVITE SDP from freeswitch:
>>>>
>>>> v=0.
>>>> o=SBC 1464589775 1464589776 IN IP4 <freeswitch-public-ip>.
>>>> s=SBC.
>>>> c=IN IP4 <freeswitch-public-ip>.
>>>> t=0 0.
>>>> *m=audio 22284 RTP/SAVP 8 0 9 3 18.*
>>>> a=rtpmap:8 PCMA/8000.
>>>> a=rtpmap:0 PCMU/8000.
>>>> a=rtpmap:9 G722/8000.
>>>> a=rtpmap:3 GSM/8000.
>>>> a=rtpmap:18 G729/8000.
>>>> a=rtcp:22285 IN IP4 <freeswitch-public-ip>.
>>>> a=crypto:1 AEAD_AES_256_GCM_8 
>>>> inline:Q/O+iUzZ9AZLcrADA7w/XcpkmuW6yArr1vsaWc0coTyWUIRLX2qfCA7XDAs.
>>>> a=crypto:2 AEAD_AES_128_GCM_8 
>>>> inline:mBb/SmBuSQNSm8lC5giUfZMCv0ZYINvsQiX2sw.
>>>> a=crypto:3 AES_CM_256_HMAC_SHA1_80 
>>>> inline:ahOTNSsdmHLIOBqvUyGyNjd+gDfAE/+jA6w7XwzGWxBMjtzf5akXNvM/OGy0jQ.
>>>> a=crypto:4 AES_CM_192_HMAC_SHA1_80 
>>>> inline:k7002pXV/SUT7JHhZTYaMV5keUTp1EP57M4rcNZnAoZZzsceTXA.
>>>> a=crypto:5 AES_CM_128_HMAC_SHA1_80 
>>>> inline:bHcV4E/OnzMkNeaFplPWt4RELILYZeGlifnJNlRV.
>>>> a=crypto:6 AES_CM_256_HMAC_SHA1_32 
>>>> inline:zJkowU1tc5rQR5BPpg2m3eE97ZqXLFFJc1Agh89XuZynPFMrXhO266+eMZCd2A.
>>>> a=crypto:7 AES_CM_192_HMAC_SHA1_32 
>>>> inline:2DwGI3UAPsLFBS5sJBsc+pzEsITQwDHCvB0u7pK/XEE3G8swsrw.
>>>> a=crypto:8 AES_CM_128_HMAC_SHA1_32 
>>>> inline:Jen6Gm56Z+4RTRmfpXTpSpoG5bduyythl1j21a15.
>>>> a=crypto:9 AES_CM_128_NULL_AUTH 
>>>> inline:Nn83liGQY7eY8LGs9qz/7EoPiHOpRgwY8H7Ts/b5.
>>>> a=ptime:20.
>>>> *m=audio 22284 RTP/AVP 8 0 9 3 18.*
>>>> a=rtpmap:8 PCMA/8000.
>>>> a=rtpmap:0 PCMU/8000.
>>>> a=rtpmap:9 G722/8000.
>>>> a=rtpmap:3 GSM/8000.
>>>> a=rtpmap:18 G729/8000.
>>>> a=rtcp:22285 IN IP4 <freeswitch-public-ip>.
>>>> a=ptime:20.
>>>>
>>>>
>>>>
>>>>
>>>> INVITE SDP from RTPengine :
>>>>
>>>> v=0
>>>> o=SBC 1464589775 1464589776 IN IP4 212.232.17.66
>>>> s=SBC
>>>> c=IN IP4 <rtpengine-public-ip>
>>>> t=0 0
>>>> *m=audio 31836 RTP/SAVPF 8 0 9 3 18*
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:3 GSM/8000
>>>> a=rtpmap:18 G729/8000
>>>> a=ptime:20
>>>> a=sendrecv
>>>> a=rtcp:31837
>>>> a=setup:actpass
>>>> a=fingerprint:sha-1 
>>>> 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F
>>>> a=ice-ufrag:NauDcjVU
>>>> a=ice-pwd:AiHk6LF4tE0GElyFMWtLeod1wH
>>>> a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 
>>>> 31836 typ host
>>>> a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 
>>>> 31837 typ host
>>>> *m=audio 31866 RTP/SAVPF 8 0 9 3 18*
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:3 GSM/8000
>>>> a=rtpmap:18 G729/8000
>>>> a=ptime:20
>>>> a=sendrecv
>>>> a=rtcp:31867
>>>> a=setup:actpass
>>>> a=fingerprint:sha-1 
>>>> 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F
>>>> a=ice-ufrag:YxCyPhQV
>>>> a=ice-pwd:2vYt8YaMhIp2DSLPeYOKDqqBX0
>>>> a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 
>>>> 31866 typ host
>>>> a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 
>>>> 31867 typ host
>>>>
>
>
>
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