[Asterisk-video] 3G to SIP problem

[Andrey Kuprianov]

Yup,

It looks like Asterisk does not support your codec. That's what your SDP says:

a=rtpmap:125 G.nX64/8000
a=rtpmap:101 /8000
a=rtpmap:100 /8000

And that's what you have in config file:

allow=alaw
allow=speex
allow=gsm

Try switching codec to one of these listed in your sip.conf.


On 11/16/06, Nikolay Milovanov <n_milovanov at mail.bg> wrote:
> Hi Guys,
>
> My Scenario is
>
> 3Gphone -> (3G network provider)->(Softswitch Cisco
> PGW)->SIP<-Asterisk<-SIP->SIP phone
>
> I am using Asterisk 1.4 beta 3.  I am calling from 3G to SIP. As I see from
> the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125
> G.nX64/8000) used by the PGW.
>
> Am I right or the problem is somewhere else? Please take a look of my config
> and the trace of the asterisk cli.
>
>
> sip.conf
>
> [general]
>
> videosupport=yes
> disallow=all                    ; First disallow all codecs
> allow=alaw                      ; Allow codecs in order of preference
> allow=h263
> allow=h263p
> allow=h261
>
> [32515901]
> type=friend
> secret=phone1
> host=dynamic
> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
> mailbox=1000 ; Mailbox for message waiting indicator
> context=sip
> videosupport=yes
> maxcallbitrate=128
> callerid= "test" <32515901>
> allow=alaw
> allow=speex
> allow=gsm
> allow=h261
> allow=h263
> allow=h263p
>
>
> Debug
>
> <--- SIP read from my.domain.com:5060 --->
> INVITE sip:32515901 at 172.18.10.100;user=phone SIP/2.0
> Via: SIP/2.0/UDP my.domain.com:5060
> ;branch=z9hG4bKterm-30-myphone-32515901-17145
> From: myphone <sip:myphone at my.domain.com;user=phone>;tag=1763495500
> To: 32515901 <sip:32515901 at 172.18.10.100;user=phone>
> Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> CSeq: 1 INVITE
> Supported: timer
> Session-Expires: 1800
> Min-SE: 1800
> Contact:  <sip:myphone at my.domain.com:5060>
> Allow:
> INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 317
>
> v=0
> c=IN IP4 85.118.195.7
> m=audio 18010 RTP/AVP 125
> a=rtpmap:125 G.nX64/8000
> a=X-pc-codec: 125 101 100
> a=rtpmap:125 G.nX64/8000
> a=rtpmap:101 /8000
> a=rtpmap:100 /8000
> a=X-sqn:0
> a=X-cap: 1 audio RTP/AVP 100
> a=X-cpar: a=rtpmap:100 X-NSE/8000
> a=X-cpar: a=fmtp:100 192-194,200-202
> a=X-cap: 2 image udptl t38
>
> <------------->
> --- (14 headers 13 lines) ---
> Using INVITE request as basis request -
> 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> Found peer 'test'
> Found RTP audio format 125
> Peer audio RTP is at port 85.118.195.7:18010
> Found description format G.nX64 for ID 125
> Found description format G.nX64 for ID 125
> Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0
> (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
> (nothing), combined - 0x0 (nothing)
> [Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible
> codecs, not accepting this offer!
>
> <--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP my.domain.com:5060
> ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=my.domain.com
> From: myphone <sip:myphone at my.domain.com;user=phone>;tag=1763495500
> To: 32515901 <sip:32515901 at 172.18.10.100;user=phone>;tag=as11b984c5
> Call-ID: 3f7a5361-3057ba40-5f7fc171-25 at my.domain.com
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
> Appreciate any help,
>
> Niko
>
>
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